$(INSTALL) -m 644 $$x "$(DESTDIR)$(ASTDATADIR)/rest-api" ; \
done
-ifeq ($(GREP),)
-else ifeq ($(GREP),:)
-else
- XML_core_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
-endif
-
+DOC_MOD_SUBDIRS := $(filter-out third-party,$(MOD_SUBDIRS))
+XML_core_en_US := $(shell build_tools/make_xml_documentation --command=print_dependencies --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)")
+# core-en_US.xml is the normal documentation created with asterisk builds.
doc/core-en_US.xml: makeopts .lastclean $(XML_core_en_US)
- @printf "Building Documentation For: "
- @echo "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" > $@
- @echo "<!DOCTYPE docs SYSTEM \"appdocsxml.dtd\">" >> $@
- @echo "<?xml-stylesheet type=\"text/xsl\" href=\"appdocsxml.xslt\"?>" >> $@
- @echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
- @for x in $(MOD_SUBDIRS); do \
- printf "$$x " ; \
- for i in `find $$x -name '*.c'`; do \
- MODULEINFO=$$($(AWK) -f build_tools/get_moduleinfo $$i) ; \
- if [ -n "$$MODULEINFO" ] ; \
- then \
- echo "<module language=\"en_US\" name=\"`$(BASENAME) $$i .c`\">" >> $@ ; \
- echo "$$MODULEINFO" >> $@ ; \
- echo "</module>" >> $@ ; \
- fi ; \
- $(AWK) -f build_tools/get_documentation $$i >> $@ ; \
- done ; \
- done
- @echo
- @echo "</docs>" >> $@
+ @build_tools/make_xml_documentation --command=create_xml --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)" \
+ --with-moduleinfo --validate --output-file=$@
-ifeq ($(GREP),)
-else ifeq ($(GREP),:)
-else
- XML_full_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
-endif
-
-doc/full-en_US.xml: makeopts .lastclean $(XML_full_en_US)
+# The full-en_US.xml target is only called by the wiki documentation generation process
+# and does special post-processing in preparation for uploading to the wiki.
+# It creates full-en_US.xml but then re-creates core-en_US.xml as well.
+doc/full-en_US.xml: makeopts .lastclean $(XML_core_en_US)
ifeq ($(PYTHON),:)
@echo "--------------------------------------------------------------------------"
@echo "--- Please install python to build full documentation ---"
@echo "--------------------------------------------------------------------------"
else
- @printf "Building Documentation For: "
- @echo "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" > $@
- @echo "<!DOCTYPE docs SYSTEM \"appdocsxml.dtd\">" >> $@
- @echo "<?xml-stylesheet type=\"text/xsl\" href=\"appdocsxml.xslt\"?>" >> $@
- @echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
- @for x in $(filter-out third-party,$(MOD_SUBDIRS)); do \
- printf "$$x " ; \
- for i in `find $$x -name '*.c'`; do \
- $(PYTHON) build_tools/get_documentation.py < $$i >> $@ ; \
- done ; \
- done
- @echo
- @echo "</docs>" >> $@
- @$(PYTHON) build_tools/post_process_documentation.py -i $@ -o "doc/core-en_US.xml"
+ @build_tools/make_xml_documentation --command=create_xml --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)" \
+ --for-wiki --validate --output-file=$@ --core-output-file=./doc/core-en_US.xml
endif
validate-docs: doc/core-en_US.xml
--- /dev/null
+#!/bin/sh
+PROGNAME="${0##*/}"
+
+if [ "$1" = "-h" ] || [ "$1" = "--help" ] ; then
+ cat <<-EOF
+ Usage: ${PROGNAME}: [ <input_file> ] [ <output_file> ]
+
+ This script takes an Asterisk makeopts file, or any file containing
+ "make" style variable assignments, and converts it into a format
+ that can be directly 'sourced' by shell scripts.
+
+ * Any spaces around the equals sign are removed.
+ * The variable value is quoted.
+ * The "make" "or" command is evaluated.
+
+ Both input and output files are optional and will default to
+ stdin and stdout respectively.
+
+ NOTE: This script relies on NO external commands and only POSIX
+ constructs. It should be runnable by any shell.
+ EOF
+ exit 1
+fi
+
+input_file="/dev/stdin"
+if [ "$1" != "" ] ; then
+ input_file="$1"
+fi
+
+output_file="/dev/stdout"
+if [ "$2" != "" ] ; then
+ output_file="$2"
+fi
+
+# orfunc is a code fragment to be added to the outp[ut file.
+# We don't WANT the variables evaluated.
+# shellcheck disable=SC2016
+orfunc='or (){ before="${1%,*}" ; after="${1#*,}" ; if [ "$before" = "" ] ; then echo "${after}" ; else echo "${before}" ; fi ; }'
+echo "${orfunc}" >"${output_file}"
+
+while read -r LINE ; do
+ var="${LINE%%=*}"
+ if [ "${var}" != "" ] ; then
+ val="${LINE#*=}"
+ if [ "${val}" != "${var}" ] ; then
+ if [ "${val%% *}" = "" ] ; then
+ echo "${var% *}=\"${val#* }\""
+ else
+ echo "${var% *}=\"${val}\""
+ fi
+ fi
+ fi
+done <"${input_file}" >>"${output_file}"
+
--- /dev/null
+#!/bin/sh
+# The GREP, SED, FIND, etc variables are all set at run time from
+# makeopts.
+# shellcheck disable=SC2154
+
+PROGNAME="${0##*/}"
+PROGDIR="${0%/*}"
+
+# Fail on errors
+set -e
+
+usage() {
+cat <<-EOF
+ Usage: ${PROGNAME} --command=(create_xml | print_dependencies)
+ --source-tree=<asterisk_source_tree> [ --mod-subdirs=<subdir_search_list> ]
+ [ --with-moduleinfo ] [--for-wiki ] [ --validate ]
+ [ --output-file=<output_xml_file> ]
+ [ --core-output-file=<core_output_xml_file> ]
+
+ command:
+ print_dependencies: Print the source files that have documentation
+ for use by "make" as dependencies.
+ create_xml: Create the actual XML output file.
+
+ source-tree: The path to the Asterisk source tree.
+
+ mod-subdirs: A quoted, space-separated list of module sub-directories
+ to search for documentation. Defaults to
+ "channels pbx apps codecs formats cdr cel bridges funcs tests main res addons"
+
+ with-moduleinfo: Include the "MODULEINFO" block from source files.
+ Default is to not include MODULEINFO
+
+ for-wiki: Perform special post processing for wiki documentation.
+ This creates two output files and therefore needs both
+ <output-file> and <core-output-file>.
+ Default is to not perform wiki post-processing.
+
+ validate: Run xmllint or xmlstarlet to validate output-file.
+
+ output-file: The XML file to write to if the command was
+ "create_xml".
+
+ core-output-file: The additional XML file to write to if the command was
+ "create_xml" with "for-wiki".
+
+EOF
+}
+
+with_moduleinfo=0
+for_wiki=0
+validate=0
+command=""
+mod_subdirs="channels pbx apps codecs formats cdr cel bridges funcs tests main res addons"
+source_tree=""
+output_file=""
+core_output_file=""
+
+for arg in "$@" ; do
+ case ${arg} in
+ --for-wiki)
+ for_wiki=1
+ ;;
+ --with-moduleinfo)
+ with_moduleinfo=1
+ ;;
+ --validate)
+ validate=1
+ ;;
+ --command=*)
+ command=${arg#*=}
+ ;;
+ --source-tree=*)
+ source_tree=${arg#*=}
+ ;;
+ --mod-subdirs=*)
+ mod_subdirs="${arg#*=}"
+ ;;
+ --output-file=*)
+ output_file=${arg#*=}
+ ;;
+ --core-output-file=*)
+ core_output_file=${arg#*=}
+ ;;
+ -h|--help)
+ usage
+ exit 0
+ ;;
+ *)
+ echo "unknown option '${arg}'."
+ usage
+ exit 1
+ ;;
+ esac
+done
+
+if [ "${command}" = "" ] ; then
+ echo "No command specified"
+ usage
+ exit 1
+fi
+
+if [ "${source_tree}" = "" ] ; then
+ echo "No source-tree specified"
+ usage
+ exit 1;
+fi
+
+if [ ! -d "${source_tree}" ] ; then
+ echo "Asterisk source tree '${source_tree}' doesn't exist."
+ exit 1
+fi
+
+if [ ! -f "${source_tree}/Makefile" ] ; then
+ echo "There's no 'Makefile' in '${source_tree}'."
+ exit 1
+fi
+
+if [ ! -f "${source_tree}/makeopts" ] ; then
+ echo "There's no 'makeopts' in '${source_tree}'. Maybe you need to run ./configure?"
+ exit 1
+fi
+
+# This will get the paths to the utilities we need, all
+# of which will be in makeopts. We need to convert the
+# format so it's sourceable.
+tmpname="/tmp/ast_makeopts.$$.env"
+trap 'rm "$tmpname" >/dev/null 2>&1' INT QUIT TERM EXIT
+"${PROGDIR}/get_sourceable_makeopts" "${source_tree}/makeopts" >"${tmpname}"
+# The file to be sourced is generated at run time and can't be checked.
+# shellcheck disable=SC1090
+. "${tmpname}"
+rm "${tmpname}" > /dev/null 2>&1 || :
+trap - INT QUIT TERM EXIT
+
+# Make sure we have everything we need.
+for c in GREP FIND AWK DIRNAME BASENAME SED CAT ; do
+ bin=$(eval "echo \${${c}}")
+ if [ "${bin}" = "" ] ; then
+ echo "The '${c}' utility was not found."
+ exit 1
+ fi
+done
+
+if [ "${for_wiki}" -eq "1" ] || [ "${validate}" -eq "1" ]; then
+ if [ "${XMLLINT}${XMLSTARLET}" = "::" ] ; then
+ echo "Either xmllint or xmlstarlet is required for wiki post-processing or validation."
+ exit 1
+ fi
+fi
+
+if [ "${command}" = "print_dependencies" ] ; then
+ for subdir in ${mod_subdirs} ; do
+ subpath="${source_tree}/${subdir}"
+ # We WANT word splitting in the following line.
+ # shellcheck disable=SC2046
+ ${GREP} -l -E '(language="en_US"|appdocsxml.dtd)' $(${FIND} "${subpath}" -name '*.c' -or -name '*.cc' -or -name '*.xml') || :
+ done
+ exit
+fi
+
+if [ "${command}" != "create_xml" ] ; then
+ echo "Command '${command}' is invalid."
+ usage
+ exit 1
+fi
+
+if [ "${output_file}" = "" ] ; then
+ echo "output-file is required for command '${command}'."
+ usage
+ exit 1;
+fi
+
+output_dir=$(${DIRNAME} "${output_file}")
+if [ ! -d "${output_dir}" ] ; then
+ echo "output destination directory '${output_dir}' doesn't exist."
+ exit 1
+fi
+
+if [ "${for_wiki}" -eq "1" ] && [ "${core_output_file}" = "" ] ; then
+ echo "core-output-file is required for command '${command}' and 'for-wiki'."
+ usage
+ exit 1;
+fi
+
+core_output_dir=$(${DIRNAME} "${core_output_file}")
+if [ ! -d "${core_output_dir}" ] ; then
+ echo "core destination directory '${core_output_dir}' doesn't exist."
+ exit 1
+fi
+
+${CAT} > "${output_file}" <<-EOF
+ <?xml version="1.0" encoding="UTF-8"?>
+ <!DOCTYPE docs SYSTEM "appdocsxml.dtd">
+ <?xml-stylesheet type="text/xsl" href="appdocsxml.xslt"?>
+ <docs xmlns:xi="http://www.w3.org/2001/XInclude">
+EOF
+
+printf "Building Documentation For: "
+
+for subdir in ${mod_subdirs} ; do
+ printf "%s " "${subdir}"
+ subdir_path="${source_tree}/${subdir}"
+ for i in $(${FIND} "${subdir_path}" -name '*.c' -or -name '*.cc'); do
+ if [ "${with_moduleinfo}" -eq "1" ] ; then
+ MODULEINFO=$(${AWK} -f "${source_tree}/build_tools/get_moduleinfo" "${i}")
+ if [ "${MODULEINFO}" != "" ] ; then
+ ${CAT} >> "${output_file}" <<-EOF
+ <module language="en_US" name="$(${BASENAME} "${i}" .c)">
+ ${MODULEINFO}
+ </module>
+ EOF
+ fi
+ fi
+ if [ "${for_wiki}" -eq "1" ] ; then
+ ${PYTHON} build_tools/get_documentation.py < "${i}" >> "${output_file}"
+ else
+ ${AWK} -f "${source_tree}/build_tools/get_documentation" "${i}" >> "${output_file}"
+ fi
+ done
+ for i in $(${FIND} "${subdir_path}" -name '*.xml') ; do
+ ${GREP} -q "appdocsxml.dtd" "${i}" || continue
+ if [ "${validate}" -eq "1" ] ;then
+ if [ "${XMLLINT}" != ":" ] ; then
+ ${XMLLINT} --dtdvalid "${source_tree}/doc/appdocsxml.dtd" --path "${source_tree}/doc" --noout "${i}" || { echo "" ; exit 1 ; }
+ else
+ ${XMLSTARLET} val -d "${source_tree}/doc/appdocsxml.dtd" "${i}" || { echo "" ; exit 1 ; }
+ fi
+ fi
+ ${SED} -r "/^\s*(<[?]xml|<.DOCTYPE|<.?docs)/d" "${i}" >> "${output_file}"
+ done
+done
+echo "</docs>" >> "${output_file}"
+echo ""
+
+if [ "${for_wiki}" -eq "1" ] ; then
+ ${PYTHON} build_tools/post_process_documentation.py -i "${output_file}" -o "${core_output_file}"
+fi
+
+if [ "${validate}" -eq "1" ] ;then
+ if [ "${XMLLINT}" != ":" ] ; then
+ ${XMLLINT} --dtdvalid "${source_tree}/doc/appdocsxml.dtd" --path "${source_tree}/doc" --noout "${output_file}" || exit 1
+ else
+ ${XMLSTARLET} val -d "${source_tree}/doc/appdocsxml.dtd" "${output_file}" || exit 1
+ fi
+fi
+
NM=@NM@
CAT=@CAT@
CUT=@CUT@
+REALPATH=@REALPATH@
+DIRNAME=@DIRNAME@
BUILD_PLATFORM=@BUILD_PLATFORM@
BUILD_CPU=@BUILD_CPU@
<support_level>core</support_level>
***/
-/*** DOCUMENTATION
- <configInfo name="res_pjsip" language="en_US">
- <synopsis>SIP Resource using PJProject</synopsis>
- <configFile name="pjsip.conf">
- <configObject name="endpoint">
- <synopsis>Endpoint</synopsis>
- <description><para>
- The <emphasis>Endpoint</emphasis> is the primary configuration object.
- It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
- dialable entries of their own. Communication with another SIP device is
- accomplished via Addresses of Record (AoRs) which have one or more
- contacts associated with them. Endpoints <emphasis>NOT</emphasis> configured to
- use a <literal>transport</literal> will default to first transport found
- in <filename>pjsip.conf</filename> that matches its type.
- </para>
- <para>Example: An Endpoint has been configured with no transport.
- When it comes time to call an AoR, PJSIP will find the
- first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
- will use the first IPv6 transport and try to send the request.
- </para>
- <para>If the anonymous endpoint identifier is in use an endpoint with the name
- "anonymous@domain" will be searched for as a last resort. If this is not found
- it will fall back to searching for "anonymous". If neither endpoints are found
- the anonymous endpoint identifier will not return an endpoint and anonymous
- calling will not be possible.
- </para>
- </description>
- <configOption name="100rel" default="yes">
- <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
- <description>
- <enumlist>
- <enum name="no" />
- <enum name="required" />
- <enum name="yes" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="aggregate_mwi" default="yes">
- <synopsis>Condense MWI notifications into a single NOTIFY.</synopsis>
- <description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
- waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
- individual NOTIFYs are sent for each mailbox.</para></description>
- </configOption>
- <configOption name="allow">
- <synopsis>Media Codec(s) to allow</synopsis>
- </configOption>
- <configOption name="allow_overlap" default="yes">
- <synopsis>Enable RFC3578 overlap dialing support.</synopsis>
- </configOption>
- <configOption name="aors">
- <synopsis>AoR(s) to be used with the endpoint</synopsis>
- <description><para>
- List of comma separated AoRs that the endpoint should be associated with.
- </para></description>
- </configOption>
- <configOption name="auth">
- <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
- <description><para>
- This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
- in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
- </para><para>
- Endpoints without an authentication object
- configured will allow connections without verification.</para>
- <note><para>
- Using the same auth section for inbound and outbound
- authentication is not recommended. There is a difference in
- meaning for an empty realm setting between inbound and outbound
- authentication uses. See the auth realm description for details.
- </para></note>
- </description>
- </configOption>
- <configOption name="callerid">
- <synopsis>CallerID information for the endpoint</synopsis>
- <description><para>
- Must be in the format <literal>Name <Number></literal>,
- or only <literal><Number></literal>.
- </para></description>
- </configOption>
- <configOption name="callerid_privacy">
- <synopsis>Default privacy level</synopsis>
- <description>
- <enumlist>
- <enum name="allowed_not_screened" />
- <enum name="allowed_passed_screen" />
- <enum name="allowed_failed_screen" />
- <enum name="allowed" />
- <enum name="prohib_not_screened" />
- <enum name="prohib_passed_screen" />
- <enum name="prohib_failed_screen" />
- <enum name="prohib" />
- <enum name="unavailable" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="callerid_tag">
- <synopsis>Internal id_tag for the endpoint</synopsis>
- </configOption>
- <configOption name="context">
- <synopsis>Dialplan context for inbound sessions</synopsis>
- </configOption>
- <configOption name="direct_media_glare_mitigation" default="none">
- <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
- <description>
- <para>
- This setting attempts to avoid creating INVITE glare scenarios
- by disabling direct media reINVITEs in one direction thereby allowing
- designated servers (according to this option) to initiate direct
- media reINVITEs without contention and significantly reducing call
- setup time.
- </para>
- <para>
- A more detailed description of how this option functions can be found on
- the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
- </para>
- <enumlist>
- <enum name="none" />
- <enum name="outgoing" />
- <enum name="incoming" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="direct_media_method" default="invite">
- <synopsis>Direct Media method type</synopsis>
- <description>
- <para>Method for setting up Direct Media between endpoints.</para>
- <enumlist>
- <enum name="invite" />
- <enum name="reinvite">
- <para>Alias for the <literal>invite</literal> value.</para>
- </enum>
- <enum name="update" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="trust_connected_line">
- <synopsis>Accept Connected Line updates from this endpoint</synopsis>
- </configOption>
- <configOption name="send_connected_line">
- <synopsis>Send Connected Line updates to this endpoint</synopsis>
- </configOption>
- <configOption name="connected_line_method" default="invite">
- <synopsis>Connected line method type</synopsis>
- <description>
- <para>Method used when updating connected line information.</para>
- <enumlist>
- <enum name="invite">
- <para>When set to <literal>invite</literal>, check the remote's Allow header and
- if UPDATE is allowed, send UPDATE instead of INVITE to avoid SDP
- renegotiation. If UPDATE is not Allowed, send INVITE.</para>
- </enum>
- <enum name="reinvite">
- <para>Alias for the <literal>invite</literal> value.</para>
- </enum>
- <enum name="update">
- <para>If set to <literal>update</literal>, send UPDATE regardless of what the remote
- Allows. </para>
- </enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="direct_media" default="yes">
- <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
- </configOption>
- <configOption name="disable_direct_media_on_nat" default="no">
- <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
- </configOption>
- <configOption name="disallow">
- <synopsis>Media Codec(s) to disallow</synopsis>
- </configOption>
- <configOption name="dtmf_mode" default="rfc4733">
- <synopsis>DTMF mode</synopsis>
- <description>
- <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
- <enumlist>
- <enum name="rfc4733">
- <para>DTMF is sent out of band of the main audio stream. This
- supercedes the older <emphasis>RFC-2833</emphasis> used within
- the older <literal>chan_sip</literal>.</para>
- </enum>
- <enum name="inband">
- <para>DTMF is sent as part of audio stream.</para>
- </enum>
- <enum name="info">
- <para>DTMF is sent as SIP INFO packets.</para>
- </enum>
- <enum name="auto">
- <para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</para>
- </enum>
- <enum name="auto_info">
- <para>DTMF is sent as RFC 4733 if the other side supports it or as SIP INFO if not.</para>
- </enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="media_address">
- <synopsis>IP address used in SDP for media handling</synopsis>
- <description><para>
- At the time of SDP creation, the IP address defined here will be used as
- the media address for individual streams in the SDP.
- </para>
- <note><para>
- Be aware that the <literal>external_media_address</literal> option, set in Transport
- configuration, can also affect the final media address used in the SDP.
- </para></note>
- </description>
- </configOption>
- <configOption name="bind_rtp_to_media_address">
- <synopsis>Bind the RTP instance to the media_address</synopsis>
- <description><para>
- If media_address is specified, this option causes the RTP instance to be bound to the
- specified ip address which causes the packets to be sent from that address.
- </para>
- </description>
- </configOption>
- <configOption name="force_rport" default="yes">
- <synopsis>Force use of return port</synopsis>
- </configOption>
- <configOption name="ice_support" default="no">
- <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
- </configOption>
- <configOption name="identify_by">
- <synopsis>Way(s) for the endpoint to be identified</synopsis>
- <description>
- <para>Endpoints and AORs can be identified in multiple ways. This
- option is a comma separated list of methods the endpoint can be
- identified.
- </para>
- <note><para>
- This option controls both how an endpoint is matched for incoming
- traffic and also how an AOR is determined if a registration
- occurs. You must list at least one method that also matches for
- AORs or the registration will fail.
- </para></note>
- <enumlist>
- <enum name="username">
- <para>Matches the endpoint or AOR ID based on the username
- and domain in the From header (or To header for AORs). If
- an exact match on both username and domain/realm fails, the
- match is retried with just the username.
- </para>
- </enum>
- <enum name="auth_username">
- <para>Matches the endpoint or AOR ID based on the username
- and realm in the Authentication header. If an exact match
- on both username and domain/realm fails, the match is
- retried with just the username.
- </para>
- <note><para>This method of identification has some security
- considerations because an Authentication header is not
- present on the first message of a dialog when digest
- authentication is used. The client can't generate it until
- the server sends the challenge in a 401 response. Since
- Asterisk normally sends a security event when an incoming
- request can't be matched to an endpoint, using this method
- requires that the security event be deferred until a request
- is received with the Authentication header and only
- generated if the username doesn't result in a match. This
- may result in a delay before an attack is recognized. You
- can control how many unmatched requests are received from
- a single ip address before a security event is generated
- using the <literal>unidentified_request</literal>
- parameters in the "global" configuration object.
- </para></note>
- </enum>
- <enum name="ip">
- <para>Matches the endpoint based on the source IP address.
- </para>
- <para>This method of identification is not configured here
- but simply allowed by this configuration option. See the
- documentation for the <literal>identify</literal>
- configuration section for more details on this method of
- endpoint identification.
- </para>
- </enum>
- <enum name="header">
- <para>Matches the endpoint based on a configured SIP header
- value.
- </para>
- <para>This method of identification is not configured here
- but simply allowed by this configuration option. See the
- documentation for the <literal>identify</literal>
- configuration section for more details on this method of
- endpoint identification.
- </para>
- </enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="redirect_method">
- <synopsis>How redirects received from an endpoint are handled</synopsis>
- <description><para>
- When a redirect is received from an endpoint there are multiple ways it can be handled.
- If this option is set to <literal>user</literal> the user portion of the redirect target
- is treated as an extension within the dialplan and dialed using a Local channel. If this option
- is set to <literal>uri_core</literal> the target URI is returned to the dialing application
- which dials it using the PJSIP channel driver and endpoint originally used. If this option is
- set to <literal>uri_pjsip</literal> the redirect occurs within chan_pjsip itself and is not exposed
- to the core at all. The <literal>uri_pjsip</literal> option has the benefit of being more efficient
- and also supporting multiple potential redirect targets. The con is that since redirection occurs
- within chan_pjsip redirecting information is not forwarded and redirection can not be
- prevented.
- </para>
- <enumlist>
- <enum name="user" />
- <enum name="uri_core" />
- <enum name="uri_pjsip" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="mailboxes">
- <synopsis>NOTIFY the endpoint when state changes for any of the specified mailboxes</synopsis>
- <description><para>
- Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state
- changes happen for any of the specified mailboxes. More than one mailbox can be
- specified with a comma-delimited string. app_voicemail mailboxes must be specified
- as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by
- external sources, such as through the res_mwi_external module, you must specify
- strings supported by the external system.
- </para><para>
- For endpoints that SUBSCRIBE for MWI, use the <literal>mailboxes</literal> option in your AOR
- configuration.
- </para></description>
- </configOption>
- <configOption name="mwi_subscribe_replaces_unsolicited">
- <synopsis>An MWI subscribe will replace sending unsolicited NOTIFYs</synopsis>
- </configOption>
- <configOption name="voicemail_extension">
- <synopsis>The voicemail extension to send in the NOTIFY Message-Account header</synopsis>
- </configOption>
- <configOption name="moh_suggest" default="default">
- <synopsis>Default Music On Hold class</synopsis>
- </configOption>
- <configOption name="outbound_auth">
- <synopsis>Authentication object(s) used for outbound requests</synopsis>
- <description><para>
- This is a comma-delimited list of <replaceable>auth</replaceable>
- sections defined in <filename>pjsip.conf</filename> used to respond
- to outbound connection authentication challenges.</para>
- <note><para>
- Using the same auth section for inbound and outbound
- authentication is not recommended. There is a difference in
- meaning for an empty realm setting between inbound and outbound
- authentication uses. See the auth realm description for details.
- </para></note>
- </description>
- </configOption>
- <configOption name="outbound_proxy">
- <synopsis>Full SIP URI of the outbound proxy used to send requests</synopsis>
- </configOption>
- <configOption name="rewrite_contact">
- <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
- <description><para>
- On inbound SIP messages from this endpoint, the Contact header or an
- appropriate Record-Route header will be changed to have the source IP
- address and port. This option does not affect outbound messages sent to
- this endpoint. This option helps servers communicate with endpoints
- that are behind NATs. This option also helps reuse reliable transport
- connections such as TCP and TLS.
- </para></description>
- </configOption>
- <configOption name="rtp_ipv6" default="no">
- <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
- </configOption>
- <configOption name="rtp_symmetric" default="no">
- <synopsis>Enforce that RTP must be symmetric</synopsis>
- </configOption>
- <configOption name="send_diversion" default="yes">
- <synopsis>Send the Diversion header, conveying the diversion
- information to the called user agent</synopsis>
- </configOption>
- <configOption name="send_history_info" default="no">
- <synopsis>Send the History-Info header, conveying the diversion
- information to the called and calling user agents</synopsis>
- </configOption>
- <configOption name="send_pai" default="no">
- <synopsis>Send the P-Asserted-Identity header</synopsis>
- </configOption>
- <configOption name="send_rpid" default="no">
- <synopsis>Send the Remote-Party-ID header</synopsis>
- </configOption>
- <configOption name="rpid_immediate" default="no">
- <synopsis>Immediately send connected line updates on unanswered incoming calls.</synopsis>
- <description>
- <para>When enabled, immediately send <emphasis>180 Ringing</emphasis>
- or <emphasis>183 Progress</emphasis> response messages to the
- caller if the connected line information is updated before
- the call is answered. This can send a <emphasis>180 Ringing</emphasis>
- response before the call has even reached the far end. The
- caller can start hearing ringback before the far end even gets
- the call. Many phones tend to grab the first connected line
- information and refuse to update the display if it changes. The
- first information is not likely to be correct if the call
- goes to an endpoint not under the control of this Asterisk
- box.</para>
- <para>When disabled, a connected line update must wait for
- another reason to send a message with the connected line
- information to the caller before the call is answered. You can
- trigger the sending of the information by using an appropriate
- dialplan application such as <emphasis>Ringing</emphasis>.</para>
- </description>
- </configOption>
- <configOption name="timers_min_se" default="90">
- <synopsis>Minimum session timers expiration period</synopsis>
- <description><para>
- Minimum session timer expiration period. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="timers" default="yes">
- <synopsis>Session timers for SIP packets</synopsis>
- <description>
- <enumlist>
- <enum name="no" />
- <enum name="yes" />
- <enum name="required" />
- <enum name="always" />
- <enum name="forced"><para>Alias of always</para></enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="timers_sess_expires" default="1800">
- <synopsis>Maximum session timer expiration period</synopsis>
- <description><para>
- Maximum session timer expiration period. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="transport">
- <synopsis>Explicit transport configuration to use</synopsis>
- <description>
- <para>This will <emphasis>force</emphasis> the endpoint to use the
- specified transport configuration to send SIP messages. You need
- to already know what kind of transport (UDP/TCP/IPv4/etc) the
- endpoint device will use.
- </para>
- <note><para>Not specifying a transport will select the first
- configured transport in <filename>pjsip.conf</filename> which is
- compatible with the URI we are trying to contact.
- </para></note>
- <warning><para>Transport configuration is not affected by reloads. In order to
- change transports, a full Asterisk restart is required</para></warning>
- </description>
- </configOption>
- <configOption name="trust_id_inbound" default="no">
- <synopsis>Accept identification information received from this endpoint</synopsis>
- <description><para>This option determines whether Asterisk will accept
- identification from the endpoint from headers such as P-Asserted-Identity
- or Remote-Party-ID header. This option applies both to calls originating from the
- endpoint and calls originating from Asterisk. If <literal>no</literal>, the
- configured Caller-ID from pjsip.conf will always be used as the identity for
- the endpoint.</para></description>
- </configOption>
- <configOption name="trust_id_outbound" default="no">
- <synopsis>Send private identification details to the endpoint.</synopsis>
- <description><para>This option determines whether res_pjsip will send private
- identification information to the endpoint. If <literal>no</literal>,
- private Caller-ID information will not be forwarded to the endpoint.
- "Private" in this case refers to any method of restricting identification.
- Example: setting <replaceable>callerid_privacy</replaceable> to any
- <literal>prohib</literal> variation.
- Example: If <replaceable>trust_id_inbound</replaceable> is set to
- <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
- header in a SIP request or response would indicate the identification
- provided in the request is private.</para></description>
- </configOption>
- <configOption name="type">
- <synopsis>Must be of type 'endpoint'.</synopsis>
- </configOption>
- <configOption name="use_ptime" default="no">
- <synopsis>Use Endpoint's requested packetization interval</synopsis>
- </configOption>
- <configOption name="use_avpf" default="no">
- <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
- endpoint.</synopsis>
- <description><para>
- If set to <literal>yes</literal>, res_pjsip will use the AVPF or SAVPF RTP
- profile for all media offers on outbound calls and media updates and will
- decline media offers not using the AVPF or SAVPF profile.
- </para><para>
- If set to <literal>no</literal>, res_pjsip will use the AVP or SAVP RTP
- profile for all media offers on outbound calls and media updates, and will
- decline media offers not using the AVP or SAVP profile.
- </para></description>
- </configOption>
- <configOption name="force_avp" default="no">
- <synopsis>Determines whether res_pjsip will use and enforce usage of AVP,
- regardless of the RTP profile in use for this endpoint.</synopsis>
- <description><para>
- If set to <literal>yes</literal>, res_pjsip will use the AVP, AVPF, SAVP, or
- SAVPF RTP profile for all media offers on outbound calls and media updates including
- those for DTLS-SRTP streams.
- </para><para>
- If set to <literal>no</literal>, res_pjsip will use the respective RTP profile
- depending on configuration.
- </para></description>
- </configOption>
- <configOption name="media_use_received_transport" default="no">
- <synopsis>Determines whether res_pjsip will use the media transport received in the
- offer SDP in the corresponding answer SDP.</synopsis>
- <description><para>
- If set to <literal>yes</literal>, res_pjsip will use the received media transport.
- </para><para>
- If set to <literal>no</literal>, res_pjsip will use the respective RTP profile
- depending on configuration.
- </para></description>
- </configOption>
- <configOption name="media_encryption" default="no">
- <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
- for this endpoint.</synopsis>
- <description>
- <enumlist>
- <enum name="no"><para>
- res_pjsip will offer no encryption and allow no encryption to be setup.
- </para></enum>
- <enum name="sdes"><para>
- res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
- transport should be used in conjunction with this option to prevent
- exposure of media encryption keys.
- </para></enum>
- <enum name="dtls"><para>
- res_pjsip will offer DTLS-SRTP setup.
- </para></enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="media_encryption_optimistic" default="no">
- <synopsis>Determines whether encryption should be used if possible but does not terminate the
- session if not achieved.</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>sdes</literal> or <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="g726_non_standard" default="no">
- <synopsis>Force g.726 to use AAL2 packing order when negotiating g.726 audio</synopsis>
- <description><para>
- When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
- packing order instead of what is recommended by RFC3551. Since this essentially
- replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
- specified in the endpoint's allowed codec list.
- </para></description>
- </configOption>
- <configOption name="inband_progress" default="no">
- <synopsis>Determines whether chan_pjsip will indicate ringing using inband
- progress.</synopsis>
- <description><para>
- If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
- when told to indicate ringing and will immediately start sending ringing
- as audio.
- </para><para>
- If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
- to indicate ringing and will NOT send it as audio.
- </para></description>
- </configOption>
- <configOption name="call_group">
- <synopsis>The numeric pickup groups for a channel.</synopsis>
- <description><para>
- Can be set to a comma separated list of numbers or ranges between the values
- of 0-63 (maximum of 64 groups).
- </para></description>
- </configOption>
- <configOption name="pickup_group">
- <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
- <description><para>
- Can be set to a comma separated list of numbers or ranges between the values
- of 0-63 (maximum of 64 groups).
- </para></description>
- </configOption>
- <configOption name="named_call_group">
- <synopsis>The named pickup groups for a channel.</synopsis>
- <description><para>
- Can be set to a comma separated list of case sensitive strings limited by
- supported line length.
- </para></description>
- </configOption>
- <configOption name="named_pickup_group">
- <synopsis>The named pickup groups that a channel can pickup.</synopsis>
- <description><para>
- Can be set to a comma separated list of case sensitive strings limited by
- supported line length.
- </para></description>
- </configOption>
- <configOption name="device_state_busy_at" default="0">
- <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
- <description><para>
- When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
- PJSIP channel driver will return busy as the device state instead of in use.
- </para></description>
- </configOption>
- <configOption name="t38_udptl" default="no">
- <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
- <description><para>
- If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
- and relayed.
- </para></description>
- </configOption>
- <configOption name="t38_udptl_ec" default="none">
- <synopsis>T.38 UDPTL error correction method</synopsis>
- <description>
- <enumlist>
- <enum name="none"><para>
- No error correction should be used.
- </para></enum>
- <enum name="fec"><para>
- Forward error correction should be used.
- </para></enum>
- <enum name="redundancy"><para>
- Redundancy error correction should be used.
- </para></enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="t38_udptl_maxdatagram" default="0">
- <synopsis>T.38 UDPTL maximum datagram size</synopsis>
- <description><para>
- This option can be set to override the maximum datagram of a remote endpoint for broken
- endpoints.
- </para></description>
- </configOption>
- <configOption name="fax_detect" default="no">
- <synopsis>Whether CNG tone detection is enabled</synopsis>
- <description><para>
- This option can be set to send the session to the fax extension when a CNG tone is
- detected.
- </para></description>
- </configOption>
- <configOption name="fax_detect_timeout">
- <synopsis>How long into a call before fax_detect is disabled for the call</synopsis>
- <description><para>
- The option determines how many seconds into a call before the
- fax_detect option is disabled for the call. Setting the value
- to zero disables the timeout.
- </para></description>
- </configOption>
- <configOption name="t38_udptl_nat" default="no">
- <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
- <description><para>
- When enabled the UDPTL stack will send UDPTL packets to the source address of
- received packets.
- </para></description>
- </configOption>
- <configOption name="t38_udptl_ipv6" default="no">
- <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
- <description><para>
- When enabled the UDPTL stack will use IPv6.
- </para></description>
- </configOption>
- <configOption name="t38_bind_udptl_to_media_address" default="no">
- <synopsis>Bind the UDPTL instance to the media_adress</synopsis>
- <description><para>
- If media_address is specified, this option causes the UDPTL instance to be bound to
- the specified ip address which causes the packets to be sent from that address.
- </para></description>
- </configOption>
- <configOption name="tone_zone">
- <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
- </configOption>
- <configOption name="language">
- <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
- </configOption>
- <configOption name="one_touch_recording" default="no">
- <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
- <see-also>
- <ref type="configOption">record_on_feature</ref>
- <ref type="configOption">record_off_feature</ref>
- </see-also>
- </configOption>
- <configOption name="record_on_feature" default="automixmon">
- <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
- <description>
- <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
- feature will be enabled for the channel. The feature designated here can be any built-in
- or dynamic feature defined in features.conf.</para>
- <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
- </description>
- <see-also>
- <ref type="configOption">one_touch_recording</ref>
- <ref type="configOption">record_off_feature</ref>
- </see-also>
- </configOption>
- <configOption name="record_off_feature" default="automixmon">
- <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
- <description>
- <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
- feature will be enabled for the channel. The feature designated here can be any built-in
- or dynamic feature defined in features.conf.</para>
- <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
- </description>
- <see-also>
- <ref type="configOption">one_touch_recording</ref>
- <ref type="configOption">record_on_feature</ref>
- </see-also>
- </configOption>
- <configOption name="rtp_engine" default="asterisk">
- <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
- </configOption>
- <configOption name="allow_transfer" default="yes">
- <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
- </configOption>
- <configOption name="user_eq_phone" default="no">
- <synopsis>Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number</synopsis>
- </configOption>
- <configOption name="moh_passthrough" default="no">
- <synopsis>Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side</synopsis>
- </configOption>
- <configOption name="sdp_owner" default="-">
- <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
- </configOption>
- <configOption name="sdp_session" default="Asterisk">
- <synopsis>String used for the SDP session (s=) line.</synopsis>
- </configOption>
- <configOption name="tos_audio">
- <synopsis>DSCP TOS bits for audio streams</synopsis>
- <description><para>
- See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
- </para></description>
- </configOption>
- <configOption name="tos_video">
- <synopsis>DSCP TOS bits for video streams</synopsis>
- <description><para>
- See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
- </para></description>
- </configOption>
- <configOption name="cos_audio">
- <synopsis>Priority for audio streams</synopsis>
- <description><para>
- See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
- </para></description>
- </configOption>
- <configOption name="cos_video">
- <synopsis>Priority for video streams</synopsis>
- <description><para>
- See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
- </para></description>
- </configOption>
- <configOption name="allow_subscribe" default="yes">
- <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
- </configOption>
- <configOption name="sub_min_expiry" default="60">
- <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
- </configOption>
- <configOption name="from_user">
- <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
- </configOption>
- <configOption name="mwi_from_user">
- <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
- </configOption>
- <configOption name="from_domain">
- <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
- </configOption>
- <configOption name="dtls_verify">
- <synopsis>Verify that the provided peer certificate is valid</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para><para>
- It can be one of the following values:
- </para><enumlist>
- <enum name="no"><para>
- meaning no verification is done.
- </para></enum>
- <enum name="fingerprint"><para>
- meaning to verify the remote fingerprint.
- </para></enum>
- <enum name="certificate"><para>
- meaning to verify the remote certificate.
- </para></enum>
- <enum name="yes"><para>
- meaning to verify both the remote fingerprint and certificate.
- </para></enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="dtls_rekey">
- <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para><para>
- If this is not set or the value provided is 0 rekeying will be disabled.
- </para></description>
- </configOption>
- <configOption name="dtls_auto_generate_cert" default="no">
- <synopsis>Whether or not to automatically generate an ephemeral X.509 certificate</synopsis>
- <description>
- <para>
- If enabled, Asterisk will generate an X.509 certificate for each DTLS session.
- This option only applies if <replaceable>media_encryption</replaceable> is set
- to <literal>dtls</literal>. This option will be automatically enabled if
- <literal>webrtc</literal> is enabled and <literal>dtls_cert_file</literal> is
- not specified.
- </para>
- </description>
- </configOption>
- <configOption name="dtls_cert_file">
- <synopsis>Path to certificate file to present to peer</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="dtls_private_key">
- <synopsis>Path to private key for certificate file</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="dtls_cipher">
- <synopsis>Cipher to use for DTLS negotiation</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para>
- <para>Many options for acceptable ciphers. See link for more:</para>
- <para>http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
- </para></description>
- </configOption>
- <configOption name="dtls_ca_file">
- <synopsis>Path to certificate authority certificate</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="dtls_ca_path">
- <synopsis>Path to a directory containing certificate authority certificates</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="dtls_setup">
- <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
- <description>
- <para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para>
- <enumlist>
- <enum name="active"><para>
- res_pjsip will make a connection to the peer.
- </para></enum>
- <enum name="passive"><para>
- res_pjsip will accept connections from the peer.
- </para></enum>
- <enum name="actpass"><para>
- res_pjsip will offer and accept connections from the peer.
- </para></enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="dtls_fingerprint">
- <synopsis>Type of hash to use for the DTLS fingerprint in the SDP.</synopsis>
- <description>
- <para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>dtls</literal>.
- </para>
- <enumlist>
- <enum name="SHA-256"></enum>
- <enum name="SHA-1"></enum>
- </enumlist>
- </description>
- </configOption>
- <configOption name="srtp_tag_32">
- <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
- <description><para>
- This option only applies if <replaceable>media_encryption</replaceable> is
- set to <literal>sdes</literal> or <literal>dtls</literal>.
- </para></description>
- </configOption>
- <configOption name="set_var">
- <synopsis>Variable set on a channel involving the endpoint.</synopsis>
- <description><para>
- When a new channel is created using the endpoint set the specified
- variable(s) on that channel. For multiple channel variables specify
- multiple 'set_var'(s).
- </para></description>
- </configOption>
- <configOption name="message_context">
- <synopsis>Context to route incoming MESSAGE requests to.</synopsis>
- <description><para>
- If specified, incoming MESSAGE requests will be routed to the indicated
- dialplan context. If no <replaceable>message_context</replaceable> is
- specified, then the <replaceable>context</replaceable> setting is used.
- </para></description>
- </configOption>
- <configOption name="accountcode">
- <synopsis>An accountcode to set automatically on any channels created for this endpoint.</synopsis>
- <description><para>
- If specified, any channel created for this endpoint will automatically
- have this accountcode set on it.
- </para></description>
- </configOption>
- <configOption name="preferred_codec_only" default="no">
- <synopsis>Respond to a SIP invite with the single most preferred codec rather than advertising all joint codec capabilities. This limits the other side's codec choice to exactly what we prefer.</synopsis>
- </configOption>
- <configOption name="rtp_keepalive">
- <synopsis>Number of seconds between RTP comfort noise keepalive packets.</synopsis>
- <description><para>
- At the specified interval, Asterisk will send an RTP comfort noise frame. This may
- be useful for situations where Asterisk is behind a NAT or firewall and must keep
- a hole open in order to allow for media to arrive at Asterisk.
- </para></description>
- </configOption>
- <configOption name="rtp_timeout" default="0">
- <synopsis>Maximum number of seconds without receiving RTP (while off hold) before terminating call.</synopsis>
- <description><para>
- This option configures the number of seconds without RTP (while off hold) before
- considering a channel as dead. When the number of seconds is reached the underlying
- channel is hung up. By default this option is set to 0, which means do not check.
- </para></description>
- </configOption>
- <configOption name="rtp_timeout_hold" default="0">
- <synopsis>Maximum number of seconds without receiving RTP (while on hold) before terminating call.</synopsis>
- <description><para>
- This option configures the number of seconds without RTP (while on hold) before
- considering a channel as dead. When the number of seconds is reached the underlying
- channel is hung up. By default this option is set to 0, which means do not check.
- </para></description>
- </configOption>
- <configOption name="acl">
- <synopsis>List of IP ACL section names in acl.conf</synopsis>
- <description><para>
- This matches sections configured in <literal>acl.conf</literal>. The value is
- defined as a list of comma-delimited section names.
- </para></description>
- </configOption>
- <configOption name="deny">
- <synopsis>List of IP addresses to deny access from</synopsis>
- <description><para>
- The value is a comma-delimited list of IP addresses. IP addresses may
- have a subnet mask appended. The subnet mask may be written in either
- CIDR or dotted-decimal notation. Separate the IP address and subnet
- mask with a slash ('/')
- </para></description>
- </configOption>
- <configOption name="permit">
- <synopsis>List of IP addresses to permit access from</synopsis>
- <description><para>
- The value is a comma-delimited list of IP addresses. IP addresses may
- have a subnet mask appended. The subnet mask may be written in either
- CIDR or dotted-decimal notation. Separate the IP address and subnet
- mask with a slash ('/')
- </para></description>
- </configOption>
- <configOption name="contact_acl">
- <synopsis>List of Contact ACL section names in acl.conf</synopsis>
- <description><para>
- This matches sections configured in <literal>acl.conf</literal>. The value is
- defined as a list of comma-delimited section names.
- </para></description>
- </configOption>
- <configOption name="contact_deny">
- <synopsis>List of Contact header addresses to deny</synopsis>
- <description><para>
- The value is a comma-delimited list of IP addresses. IP addresses may
- have a subnet mask appended. The subnet mask may be written in either
- CIDR or dotted-decimal notation. Separate the IP address and subnet
- mask with a slash ('/')
- </para></description>
- </configOption>
- <configOption name="contact_permit">
- <synopsis>List of Contact header addresses to permit</synopsis>
- <description><para>
- The value is a comma-delimited list of IP addresses. IP addresses may
- have a subnet mask appended. The subnet mask may be written in either
- CIDR or dotted-decimal notation. Separate the IP address and subnet
- mask with a slash ('/')
- </para></description>
- </configOption>
- <configOption name="subscribe_context">
- <synopsis>Context for incoming MESSAGE requests.</synopsis>
- <description><para>
- If specified, incoming SUBSCRIBE requests will be searched for the matching
- extension in the indicated context.
- If no <replaceable>subscribe_context</replaceable> is specified,
- then the <replaceable>context</replaceable> setting is used.
- </para></description>
- </configOption>
- <configOption name="contact_user" default="">
- <synopsis>Force the user on the outgoing Contact header to this value.</synopsis>
- <description><para>
- On outbound requests, force the user portion of the Contact header to this value.
- </para></description>
- </configOption>
- <configOption name="asymmetric_rtp_codec" default="no">
- <synopsis>Allow the sending and receiving RTP codec to differ</synopsis>
- <description><para>
- When set to "yes" the codec in use for sending will be allowed to differ from
- that of the received one. PJSIP will not automatically switch the sending one
- to the receiving one.
- </para></description>
- </configOption>
- <configOption name="rtcp_mux" default="no">
- <synopsis>Enable RFC 5761 RTCP multiplexing on the RTP port</synopsis>
- <description><para>
- With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux"
- attribute on all media streams. This will result in RTP and RTCP being sent and received
- on the same port. This shifts the demultiplexing logic to the application rather than
- the transport layer. This option is useful when interoperating with WebRTC endpoints
- since they mandate this option's use.
- </para></description>
- </configOption>
- <configOption name="refer_blind_progress" default="yes">
- <synopsis>Whether to notifies all the progress details on blind transfer</synopsis>
- <description><para>
- Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK"
- after REFER has been accepted. If set to <literal>no</literal> then asterisk
- will not send the progress details, but immediately will send "200 OK".
- </para></description>
- </configOption>
- <configOption name="notify_early_inuse_ringing" default="no">
- <synopsis>Whether to notifies dialog-info 'early' on InUse&Ringing state</synopsis>
- <description><para>
- Control whether dialog-info subscriptions get 'early' state
- on Ringing when already INUSE.
- </para></description>
- </configOption>
- <configOption name="max_audio_streams" default="1">
- <synopsis>The maximum number of allowed audio streams for the endpoint</synopsis>
- <description><para>
- This option enforces a limit on the maximum simultaneous negotiated audio
- streams allowed for the endpoint.
- </para></description>
- </configOption>
- <configOption name="max_video_streams" default="1">
- <synopsis>The maximum number of allowed video streams for the endpoint</synopsis>
- <description><para>
- This option enforces a limit on the maximum simultaneous negotiated video
- streams allowed for the endpoint.
- </para></description>
- </configOption>
- <configOption name="bundle" default="no">
- <synopsis>Enable RTP bundling</synopsis>
- <description><para>
- With this option enabled, Asterisk will attempt to negotiate the use of bundle.
- If negotiated this will result in multiple RTP streams being carried over the same
- underlying transport. Note that enabling bundle will also enable the rtcp_mux option.
- </para></description>
- </configOption>
- <configOption name="webrtc" default="no">
- <synopsis>Defaults and enables some options that are relevant to WebRTC</synopsis>
- <description><para>
- When set to "yes" this also enables the following values that are needed in
- order for basic WebRTC support to work: rtcp_mux, use_avpf, ice_support, and
- use_received_transport. The following configuration settings also get defaulted
- as follows:</para>
- <para>media_encryption=dtls</para>
- <para>dtls_auto_generate_cert=yes (if dtls_cert_file is not set)</para>
- <para>dtls_verify=fingerprint</para>
- <para>dtls_setup=actpass</para>
- </description>
- </configOption>
- <configOption name="incoming_mwi_mailbox">
- <synopsis>Mailbox name to use when incoming MWI NOTIFYs are received</synopsis>
- <description><para>
- If an MWI NOTIFY is received <emphasis>from</emphasis> this endpoint,
- this mailbox will be used when notifying other modules of MWI status
- changes. If not set, incoming MWI NOTIFYs are ignored.
- </para></description>
- </configOption>
- <configOption name="follow_early_media_fork">
- <synopsis>Follow SDP forked media when To tag is different</synopsis>
- <description><para>
- On outgoing calls, if the UAS responds with different SDP attributes
- on subsequent 18X or 2XX responses (such as a port update) AND the
- To tag on the subsequent response is different than that on the previous
- one, follow it. This usually happens when the INVITE is forked to multiple
- UASs and more than one sends an SDP answer.
- </para>
- <note><para>
- This option must also be enabled in the <literal>system</literal>
- section for it to take effect here.
- </para></note>
- </description>
- </configOption>
- <configOption name="accept_multiple_sdp_answers" default="no">
- <synopsis>Accept multiple SDP answers on non-100rel responses</synopsis>
- <description><para>
- On outgoing calls, if the UAS responds with different SDP attributes
- on non-100rel 18X or 2XX responses (such as a port update) AND the
- To tag on the subsequent response is the same as that on the previous one,
- process the updated SDP. This can happen when the UAS needs to change ports
- for some reason such as using a separate port for custom ringback.
- </para>
- <note><para>
- This option must also be enabled in the <literal>system</literal>
- section for it to take effect here.
- </para></note>
- </description>
- </configOption>
- <configOption name="suppress_q850_reason_headers" default="no">
- <synopsis>Suppress Q.850 Reason headers for this endpoint</synopsis>
- <description><para>
- Some devices can't accept multiple Reason headers and get confused
- when both 'SIP' and 'Q.850' Reason headers are received. This
- option allows the 'Q.850' Reason header to be suppressed.</para>
- </description>
- </configOption>
- <configOption name="ignore_183_without_sdp" default="no">
- <synopsis>Do not forward 183 when it doesn't contain SDP</synopsis>
- <description><para>
- Certain SS7 internetworking scenarios can result in a 183
- to be generated for reasons other than early media. Forwarding
- this 183 can cause loss of ringback tone. This flag emulates
- the behavior of chan_sip and prevents these 183 responses from
- being forwarded.</para>
- </description>
- </configOption>
- <configOption name="stir_shaken" default="no">
- <synopsis>Enable STIR/SHAKEN support on this endpoint</synopsis>
- <description><para>
- Enable STIR/SHAKEN support on this endpoint. On incoming INVITEs,
- the Identity header will be checked for validity. On outgoing
- INVITEs, an Identity header will be added.</para>
- </description>
- </configOption>
- <configOption name="allow_unauthenticated_options" default="no">
- <synopsis>Skip authentication when receiving OPTIONS requests</synopsis>
- <description><para>
- RFC 3261 says that the response to an OPTIONS request MUST be the
- same had the request been an INVITE. Some UAs use OPTIONS requests
- like a 'ping' and the expectation is that they will return a
- 200 OK.</para>
- <para>Enabling <literal>allow_unauthenticated_options</literal>
- will skip authentication of OPTIONS requests for the given
- endpoint.</para>
- <para>There are security implications to enabling this setting as
- it can allow information disclosure to occur - specifically, if
- enabled, an external party could enumerate and find the endpoint
- name by sending OPTIONS requests and examining the
- responses.</para>
- </description>
- </configOption>
- </configObject>
- <configObject name="auth">
- <synopsis>Authentication type</synopsis>
- <description><para>
- Authentication objects hold the authentication information for use
- by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
- This also allows for multiple objects to use a single auth object. See
- the <literal>auth_type</literal> config option for password style choices.
- </para></description>
- <configOption name="auth_type" default="userpass">
- <synopsis>Authentication type</synopsis>
- <description><para>
- This option specifies which of the password style config options should be read
- when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
- then we'll read from the 'password' option. For <literal>md5</literal> we'll read
- from 'md5_cred'. The following values are valid:
- </para>
- <enumlist>
- <enum name="md5"/>
- <enum name="userpass"/>
- </enumlist>
- <para>
- </para>
- <note>
- <para>
- This setting only describes whether the password is in
- plain text or has been pre-hashed with MD5. It doesn't describe
- the acceptable digest algorithms we'll accept in a received
- challenge.
- </para>
- </note>
- </description>
- </configOption>
- <configOption name="nonce_lifetime" default="32">
- <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
- </configOption>
- <configOption name="md5_cred" default="">
- <synopsis>MD5 Hash used for authentication.</synopsis>
- <description><para>
- Only used when auth_type is <literal>md5</literal>.
- As an alternative to specifying a plain text password,
- you can hash the username, realm and password
- together one time and place the hash value here.
- The input to the hash function must be in the
- following format:
- </para>
- <para>
- </para>
- <para>
- <username>:<realm>:<password>
- </para>
- <para>
- </para>
- <para>
- For incoming authentication (asterisk is the server),
- the realm must match either the realm set in this object
- or the <variable>default_realm</variable> set in in the
- <replaceable>global</replaceable> object.
- </para>
- <para>
- </para>
- <para>
- For outgoing authentication (asterisk is the UAC),
- the realm must match what the server will be sending
- in their WWW-Authenticate header. It can't be blank
- unless you expect the server to be sending a blank
- realm in the header. You can't use pre-hashed
- passwords with a wildcard auth object.
- You can generate the hash with the following shell
- command:
- </para>
- <para>
- </para>
- <para>
- $ echo -n "myname:myrealm:mypassword" | md5sum
- </para>
- <para>
- </para>
- <para>
- Note the '-n'. You don't want a newline to be part
- of the hash.
- </para></description>
- </configOption>
- <configOption name="password">
- <synopsis>Plain text password used for authentication.</synopsis>
- <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
- </configOption>
- <configOption name="realm" default="">
- <synopsis>SIP realm for endpoint</synopsis>
- <description><para>
- For incoming authentication (asterisk is the UAS),
- this is the realm to be sent on WWW-Authenticate
- headers. If not specified, the <replaceable>global</replaceable>
- object's <variable>default_realm</variable> will be used.
- </para>
- <para>
- </para>
- <para>
- For outgoing authentication (asterisk is the UAC), this
- must either be the realm the server is expected to send,
- or left blank or contain a single '*' to automatically
- use the realm sent by the server. If you have multiple
- auth objects for an endpoint, the realm is also used to
- match the auth object to the realm the server sent.
- </para>
- <para>
- </para>
- <note>
- <para>
- Using the same auth section for inbound and outbound
- authentication is not recommended. There is a difference in
- meaning for an empty realm setting between inbound and outbound
- authentication uses.
- </para>
- </note>
- <para>
- </para>
- <note>
- <para>
- If more than one auth object with the same realm or
- more than one wildcard auth object associated to
- an endpoint, we can only use the first one of
- each defined on the endpoint.
- </para>
- </note>
- </description>
- </configOption>
- <configOption name="type">
- <synopsis>Must be 'auth'</synopsis>
- </configOption>
- <configOption name="username">
- <synopsis>Username to use for account</synopsis>
- </configOption>
- </configObject>
- <configObject name="domain_alias">
- <synopsis>Domain Alias</synopsis>
- <description><para>
- Signifies that a domain is an alias. If the domain on a session is
- not found to match an AoR then this object is used to see if we have
- an alias for the AoR to which the endpoint is binding. This objects
- name as defined in configuration should be the domain alias and a
- config option is provided to specify the domain to be aliased.
- </para></description>
- <configOption name="type">
- <synopsis>Must be of type 'domain_alias'.</synopsis>
- </configOption>
- <configOption name="domain">
- <synopsis>Domain to be aliased</synopsis>
- </configOption>
- </configObject>
- <configObject name="transport">
- <synopsis>SIP Transport</synopsis>
- <description><para>
- <emphasis>Transports</emphasis>
- </para>
- <para>There are different transports and protocol derivatives
- supported by <literal>res_pjsip</literal>. They are in order of
- preference: UDP, TCP, and WebSocket (WS).</para>
- <note><para>Changes to transport configuration in pjsip.conf will only be
- effected on a complete restart of Asterisk. A module reload
- will not suffice.</para></note>
- </description>
- <configOption name="async_operations" default="1">
- <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
- </configOption>
- <configOption name="bind">
- <synopsis>IP Address and optional port to bind to for this transport</synopsis>
- </configOption>
- <configOption name="ca_list_file">
- <synopsis>File containing a list of certificates to read (TLS ONLY, not WSS)</synopsis>
- </configOption>
- <configOption name="ca_list_path">
- <synopsis>Path to directory containing a list of certificates to read (TLS ONLY, not WSS)</synopsis>
- </configOption>
- <configOption name="cert_file">
- <synopsis>Certificate file for endpoint (TLS ONLY, not WSS)</synopsis>
- <description><para>
- A path to a .crt or .pem file can be provided. However, only
- the certificate is read from the file, not the private key.
- The <literal>priv_key_file</literal> option must supply a
- matching key file.
- </para></description>
- </configOption>
- <configOption name="cipher">
- <synopsis>Preferred cryptography cipher names (TLS ONLY, not WSS)</synopsis>
- <description>
- <para>Comma separated list of cipher names or numeric equivalents.
- Numeric equivalents can be either decimal or hexadecimal (0xX).
- </para>
- <para>There are many cipher names. Use the CLI command
- <literal>pjsip list ciphers</literal> to see a list of cipher
- names available for your installation. See link for more:</para>
- <para>http://www.openssl.org/docs/apps/ciphers.html#CIPHER_SUITE_NAMES
- </para>
- </description>
- </configOption>
- <configOption name="domain">
- <synopsis>Domain the transport comes from</synopsis>
- </configOption>
- <configOption name="external_media_address">
- <synopsis>External IP address to use in RTP handling</synopsis>
- <description><para>
- When a request or response is sent out, if the destination of the
- message is outside the IP network defined in the option <literal>localnet</literal>,
- and the media address in the SDP is within the localnet network, then the
- media address in the SDP will be rewritten to the value defined for
- <literal>external_media_address</literal>.
- </para></description>
- </configOption>
- <configOption name="external_signaling_address">
- <synopsis>External address for SIP signalling</synopsis>
- </configOption>
- <configOption name="external_signaling_port" default="0">
- <synopsis>External port for SIP signalling</synopsis>
- </configOption>
- <configOption name="method">
- <synopsis>Method of SSL transport (TLS ONLY, not WSS)</synopsis>
- <description>
- <enumlist>
- <enum name="default">
- <para>The default as defined by PJSIP. This is currently TLSv1, but may change with future releases.</para>
- </enum>
- <enum name="unspecified">
- <para>This option is equivalent to setting 'default'</para>
- </enum>
- <enum name="tlsv1" />
- <enum name="tlsv1_1" />
- <enum name="tlsv1_2" />
- <enum name="sslv2" />
- <enum name="sslv3" />
- <enum name="sslv23" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="local_net">
- <synopsis>Network to consider local (used for NAT purposes).</synopsis>
- <description><para>This must be in CIDR or dotted decimal format with the IP
- and mask separated with a slash ('/').</para></description>
- </configOption>
- <configOption name="password">
- <synopsis>Password required for transport</synopsis>
- </configOption>
- <configOption name="priv_key_file">
- <synopsis>Private key file (TLS ONLY, not WSS)</synopsis>
- </configOption>
- <configOption name="protocol" default="udp">
- <synopsis>Protocol to use for SIP traffic</synopsis>
- <description>
- <enumlist>
- <enum name="udp" />
- <enum name="tcp" />
- <enum name="tls" />
- <enum name="ws" />
- <enum name="wss" />
- </enumlist>
- </description>
- </configOption>
- <configOption name="require_client_cert" default="false">
- <synopsis>Require client certificate (TLS ONLY, not WSS)</synopsis>
- </configOption>
- <configOption name="type">
- <synopsis>Must be of type 'transport'.</synopsis>
- </configOption>
- <configOption name="verify_client" default="false">
- <synopsis>Require verification of client certificate (TLS ONLY, not WSS)</synopsis>
- </configOption>
- <configOption name="verify_server" default="false">
- <synopsis>Require verification of server certificate (TLS ONLY, not WSS)</synopsis>
- </configOption>
- <configOption name="tos" default="false">
- <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
- <description>
- <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
- for more information on this parameter.</para>
- <note><para>This option does not apply to the <replaceable>ws</replaceable>
- or the <replaceable>wss</replaceable> protocols.</para></note>
- </description>
- </configOption>
- <configOption name="cos" default="false">
- <synopsis>Enable COS for the signalling sent over this transport</synopsis>
- <description>
- <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
- for more information on this parameter.</para>
- <note><para>This option does not apply to the <replaceable>ws</replaceable>
- or the <replaceable>wss</replaceable> protocols.</para></note>
- </description>
- </configOption>
- <configOption name="websocket_write_timeout" default="100">
- <synopsis>The timeout (in milliseconds) to set on WebSocket connections.</synopsis>
- <description>
- <para>If a websocket connection accepts input slowly, the timeout
- for writes to it can be increased to keep it from being disconnected.
- Value is in milliseconds.</para>
- </description>
- </configOption>
- <configOption name="allow_reload" default="no">
- <synopsis>Allow this transport to be reloaded.</synopsis>
- <description>
- <para>Allow this transport to be reloaded when res_pjsip is reloaded.
- This option defaults to "no" because reloading a transport may disrupt
- in-progress calls.</para>
- </description>
- </configOption>
- <configOption name="symmetric_transport" default="no">
- <synopsis>Use the same transport for outgoing requests as incoming ones.</synopsis>
- <description>
- <para>When a request from a dynamic contact
- comes in on a transport with this option set to 'yes',
- the transport name will be saved and used for subsequent
- outgoing requests like OPTIONS, NOTIFY and INVITE. It's
- saved as a contact uri parameter named 'x-ast-txp' and will
- display with the contact uri in CLI, AMI, and ARI output.
- On the outgoing request, if a transport wasn't explicitly
- set on the endpoint AND the request URI is not a hostname,
- the saved transport will be used and the 'x-ast-txp'
- parameter stripped from the outgoing packet.
- </para>
- </description>
- </configOption>
- </configObject>
- <configObject name="contact">
- <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
- <description><para>
- Contacts are a way to hide SIP URIs from the dialplan directly.
- They are also used to make a group of contactable parties when
- in use with <literal>AoR</literal> lists.
- </para></description>
- <configOption name="type">
- <synopsis>Must be of type 'contact'.</synopsis>
- </configOption>
- <configOption name="uri">
- <synopsis>SIP URI to contact peer</synopsis>
- </configOption>
- <configOption name="expiration_time">
- <synopsis>Time to keep alive a contact</synopsis>
- <description><para>
- Time to keep alive a contact. String style specification.
- </para></description>
- </configOption>
- <configOption name="qualify_frequency" default="0">
- <synopsis>Interval at which to qualify a contact</synopsis>
- <description><para>
- Interval between attempts to qualify the contact for reachability.
- If <literal>0</literal> never qualify. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="qualify_timeout" default="3.0">
- <synopsis>Timeout for qualify</synopsis>
- <description><para>
- If the contact doesn't respond to the OPTIONS request before the timeout,
- the contact is marked unavailable.
- If <literal>0</literal> no timeout. Time in fractional seconds.
- </para></description>
- </configOption>
- <configOption name="authenticate_qualify">
- <synopsis>Authenticates a qualify challenge response if needed</synopsis>
- <description>
- <para>If true and a qualify request receives a challenge response then
- authentication is attempted before declaring the contact available.
- </para>
- <note><para>This option does nothing as we will always complete
- the challenge response authentication if the qualify request is
- challenged.
- </para></note>
- </description>
- </configOption>
- <configOption name="outbound_proxy">
- <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
- <description><para>
- If set the provided URI will be used as the outbound proxy when an
- OPTIONS request is sent to a contact for qualify purposes.
- </para></description>
- </configOption>
- <configOption name="path">
- <synopsis>Stored Path vector for use in Route headers on outgoing requests.</synopsis>
- </configOption>
- <configOption name="user_agent">
- <synopsis>User-Agent header from registration.</synopsis>
- <description><para>
- The User-Agent is automatically stored based on data present in incoming SIP
- REGISTER requests and is not intended to be configured manually.
- </para></description>
- </configOption>
- <configOption name="endpoint">
- <synopsis>Endpoint name</synopsis>
- <description><para>
- The name of the endpoint this contact belongs to
- </para></description>
- </configOption>
- <configOption name="reg_server">
- <synopsis>Asterisk Server name</synopsis>
- <description><para>
- Asterisk Server name on which SIP endpoint registered.
- </para></description>
- </configOption>
- <configOption name="via_addr">
- <synopsis>IP-address of the last Via header from registration.</synopsis>
- <description><para>
- The last Via header should contain the address of UA which sent the request.
- The IP-address of the last Via header is automatically stored based on data present
- in incoming SIP REGISTER requests and is not intended to be configured manually.
- </para></description>
- </configOption>
- <configOption name="via_port">
- <synopsis>IP-port of the last Via header from registration.</synopsis>
- <description><para>
- The IP-port of the last Via header is automatically stored based on data present
- in incoming SIP REGISTER requests and is not intended to be configured manually.
- </para></description>
- </configOption>
- <configOption name="call_id">
- <synopsis>Call-ID header from registration.</synopsis>
- <description><para>
- The Call-ID header is automatically stored based on data present
- in incoming SIP REGISTER requests and is not intended to be configured manually.
- </para></description>
- </configOption>
- <configOption name="prune_on_boot">
- <synopsis>A contact that cannot survive a restart/boot.</synopsis>
- <description><para>
- The option is set if the incoming SIP REGISTER contact is rewritten
- on a reliable transport and is not intended to be configured manually.
- </para></description>
- </configOption>
- </configObject>
- <configObject name="aor">
- <synopsis>The configuration for a location of an endpoint</synopsis>
- <description><para>
- An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
- AoRs are specified, an endpoint will not be reachable by Asterisk.
- Beyond that, an AoR has other uses within Asterisk, such as inbound
- registration.
- </para><para>
- An <literal>AoR</literal> is a way to allow dialing a group
- of <literal>Contacts</literal> that all use the same
- <literal>endpoint</literal> for calls.
- </para><para>
- This can be used as another way of grouping a list of contacts to dial
- rather than specifying them each directly when dialing via the dialplan.
- This must be used in conjunction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
- </para><para>
- Registrations: For Asterisk to match an inbound registration to an endpoint,
- the AoR object name must match the user portion of the SIP URI in the "To:"
- header of the inbound SIP registration. That will usually be equivalent
- to the "user name" set in your hard or soft phones configuration.
- </para></description>
- <configOption name="contact">
- <synopsis>Permanent contacts assigned to AoR</synopsis>
- <description><para>
- Contacts specified will be called whenever referenced
- by <literal>chan_pjsip</literal>.
- </para><para>
- Use a separate "contact=" entry for each contact required. Contacts
- are specified using a SIP URI.
- </para></description>
- </configOption>
- <configOption name="default_expiration" default="3600">
- <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
- </configOption>
- <configOption name="mailboxes">
- <synopsis>Allow subscriptions for the specified mailbox(es)</synopsis>
- <description><para>This option applies when an external entity subscribes to an AoR
- for Message Waiting Indications. The mailboxes specified will be subscribed to.
- More than one mailbox can be specified with a comma-delimited string.
- app_voicemail mailboxes must be specified as mailbox@context;
- for example: mailboxes=6001@default. For mailboxes provided by external sources,
- such as through the res_mwi_external module, you must specify strings supported by
- the external system.
- </para><para>
- For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
- endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint.
- </para></description>
- </configOption>
- <configOption name="voicemail_extension">
- <synopsis>The voicemail extension to send in the NOTIFY Message-Account header</synopsis>
- </configOption>
- <configOption name="maximum_expiration" default="7200">
- <synopsis>Maximum time to keep an AoR</synopsis>
- <description><para>
- Maximum time to keep a peer with explicit expiration. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="max_contacts" default="0">
- <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
- <description><para>
- Maximum number of contacts that can associate with this AoR. This value does
- not affect the number of contacts that can be added with the "contact" option.
- It only limits contacts added through external interaction, such as
- registration.
- </para>
- <note><para>The <replaceable>rewrite_contact</replaceable> option
- registers the source address as the contact address to help with
- NAT and reusing connection oriented transports such as TCP and
- TLS. Unfortunately, refreshing a registration may register a
- different contact address and exceed
- <replaceable>max_contacts</replaceable>. The
- <replaceable>remove_existing</replaceable> and
- <replaceable>remove_unavailable</replaceable> options can help by
- removing either the soonest to expire or unavailable contact(s) over
- <replaceable>max_contacts</replaceable> which is likely the
- old <replaceable>rewrite_contact</replaceable> contact source
- address being refreshed.
- </para></note>
- <note><para>This should be set to <literal>1</literal> and
- <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
- wish to stick with the older <literal>chan_sip</literal> behaviour.
- </para></note>
- </description>
- </configOption>
- <configOption name="minimum_expiration" default="60">
- <synopsis>Minimum keep alive time for an AoR</synopsis>
- <description><para>
- Minimum time to keep a peer with an explicit expiration. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="remove_existing" default="no">
- <synopsis>Determines whether new contacts replace existing ones.</synopsis>
- <description><para>
- On receiving a new registration to the AoR should it remove enough
- existing contacts not added or updated by the registration to
- satisfy <replaceable>max_contacts</replaceable>? Any removed
- contacts will expire the soonest.
- </para>
- <note><para>The <replaceable>rewrite_contact</replaceable> option
- registers the source address as the contact address to help with
- NAT and reusing connection oriented transports such as TCP and
- TLS. Unfortunately, refreshing a registration may register a
- different contact address and exceed
- <replaceable>max_contacts</replaceable>. The
- <replaceable>remove_existing</replaceable> option can help by
- removing the soonest to expire contact(s) over
- <replaceable>max_contacts</replaceable> which is likely the
- old <replaceable>rewrite_contact</replaceable> contact source
- address being refreshed.
- </para></note>
- <note><para>This should be set to <literal>yes</literal> and
- <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
- wish to stick with the older <literal>chan_sip</literal> behaviour.
- </para></note>
- </description>
- </configOption>
- <configOption name="remove_unavailable" default="no">
- <synopsis>Determines whether new contacts should replace unavailable ones.</synopsis>
- <description><para>
- The effect of this setting depends on the setting of
- <replaceable>remove_existing</replaceable>.</para>
- <para>If <replaceable>remove_existing</replaceable> is set to
- <literal>no</literal> (default), setting remove_unavailable to
- <literal>yes</literal> will remove only unavailable contacts that exceed
- <replaceable>max_contacts</replaceable> to allow an incoming
- REGISTER to complete sucessfully.</para>
- <para>If <replaceable>remove_existing</replaceable> is set to
- <literal>yes</literal>, setting remove_unavailable to
- <literal>yes</literal> will prioritize unavailable contacts for removal
- instead of just removing the contact that expires the soonest.</para>
- <note><para>See <replaceable>remove_existing</replaceable> and
- <replaceable>max_contacts</replaceable> for further information about how
- these 3 settings interact.
- </para></note>
- </description>
- </configOption>
- <configOption name="type">
- <synopsis>Must be of type 'aor'.</synopsis>
- </configOption>
- <configOption name="qualify_frequency" default="0">
- <synopsis>Interval at which to qualify an AoR</synopsis>
- <description><para>
- Interval between attempts to qualify the AoR for reachability.
- If <literal>0</literal> never qualify. Time in seconds.
- </para></description>
- </configOption>
- <configOption name="qualify_timeout" default="3.0">
- <synopsis>Timeout for qualify</synopsis>
- <description><para>
- If the contact doesn't respond to the OPTIONS request before the timeout,
- the contact is marked unavailable.
- If <literal>0</literal> no timeout. Time in fractional seconds.
- </para></description>
- </configOption>
- <configOption name="authenticate_qualify">
- <synopsis>Authenticates a qualify challenge response if needed</synopsis>
- <description>
- <para>If true and a qualify request receives a challenge response then
- authentication is attempted before declaring the contact available.
- </para>
- <note><para>This option does nothing as we will always complete
- the challenge response authentication if the qualify request is
- challenged.
- </para></note>
- </description>
- </configOption>
- <configOption name="outbound_proxy">
- <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
- <description><para>
- If set the provided URI will be used as the outbound proxy when an
- OPTIONS request is sent to a contact for qualify purposes.
- </para></description>
- </configOption>
- <configOption name="support_path">
- <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
- <description><para>
- When this option is enabled, the Path headers in register requests will be saved
- and its contents will be used in Route headers for outbound out-of-dialog requests
- and in Path headers for outbound 200 responses. Path support will also be indicated
- in the Supported header.
- </para></description>
- </configOption>
- </configObject>
- <configObject name="system">
- <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
- <description><para>
- The settings in this section are global. In addition to being global, the values will
- not be re-evaluated when a reload is performed. This is because the values must be set
- before the SIP stack is initialized. The only way to reset these values is to either
- restart Asterisk, or unload res_pjsip.so and then load it again.
- </para></description>
- <configOption name="timer_t1" default="500">
- <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
- <description><para>
- Timer T1 is the base for determining how long to wait before retransmitting
- requests that receive no response when using an unreliable transport (e.g. UDP).
- For more information on this timer, see RFC 3261, Section 17.1.1.1.
- </para></description>
- </configOption>
- <configOption name="timer_b" default="32000">
- <synopsis>Set transaction timer B value (milliseconds).</synopsis>
- <description><para>
- Timer B determines the maximum amount of time to wait after sending an INVITE
- request before terminating the transaction. It is recommended that this be set
- to 64 * Timer T1, but it may be set higher if desired. For more information on
- this timer, see RFC 3261, Section 17.1.1.1.
- </para></description>
- </configOption>
- <configOption name="compact_headers" default="no">
- <synopsis>Use the short forms of common SIP header names.</synopsis>
- </configOption>
- <configOption name="threadpool_initial_size" default="0">
- <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
- </configOption>
- <configOption name="threadpool_auto_increment" default="5">
- <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
- </configOption>
- <configOption name="threadpool_idle_timeout" default="60">
- <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
- </configOption>
- <configOption name="threadpool_max_size" default="0">
- <synopsis>Maximum number of threads in the res_pjsip threadpool.
- A value of 0 indicates no maximum.</synopsis>
- </configOption>
- <configOption name="disable_tcp_switch" default="yes">
- <synopsis>Disable automatic switching from UDP to TCP transports.</synopsis>
- <description><para>
- Disable automatic switching from UDP to TCP transports if outgoing
- request is too large. See RFC 3261 section 18.1.1.
- </para></description>
- </configOption>
- <configOption name="follow_early_media_fork">
- <synopsis>Follow SDP forked media when To tag is different</synopsis>
- <description><para>
- On outgoing calls, if the UAS responds with different SDP attributes
- on subsequent 18X or 2XX responses (such as a port update) AND the
- To tag on the subsequent response is different than that on the previous
- one, follow it.
- </para>
- <note><para>
- This option must also be enabled on endpoints that require
- this functionality.
- </para></note>
- </description>
- </configOption>
- <configOption name="accept_multiple_sdp_answers">
- <synopsis>Follow SDP forked media when To tag is the same</synopsis>
- <description><para>
- On outgoing calls, if the UAS responds with different SDP attributes
- on non-100rel 18X or 2XX responses (such as a port update) AND the
- To tag on the subsequent response is the same as that on the previous one,
- process the updated SDP.
- </para>
- <note><para>
- This option must also be enabled on endpoints that require
- this functionality.
- </para></note>
- </description>
- </configOption>
- <configOption name="disable_rport" default="no">
- <synopsis>Disable the use of rport in outgoing requests.</synopsis>
- <description><para>
- Remove "rport" parameter from the outgoing requests.
- </para></description>
- </configOption>
- <configOption name="type">
- <synopsis>Must be of type 'system' UNLESS the object name is 'system'.</synopsis>
- </configOption>
- </configObject>
- <configObject name="global">
- <synopsis>Options that apply globally to all SIP communications</synopsis>
- <description><para>
- The settings in this section are global. Unlike options in the <literal>system</literal>
- section, these options can be refreshed by performing a reload.
- </para></description>
- <configOption name="max_forwards" default="70">
- <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
- </configOption>
- <configOption name="keep_alive_interval" default="90">
- <synopsis>The interval (in seconds) to send keepalives to active connection-oriented transports.</synopsis>
- </configOption>
- <configOption name="contact_expiration_check_interval" default="30">
- <synopsis>The interval (in seconds) to check for expired contacts.</synopsis>
- </configOption>
- <configOption name="disable_multi_domain" default="no">
- <synopsis>Disable Multi Domain support</synopsis>
- <description><para>
- If disabled it can improve realtime performance by reducing the number of database requests.
- </para></description>
- </configOption>
- <configOption name="max_initial_qualify_time" default="0">
- <synopsis>The maximum amount of time from startup that qualifies should be attempted on all contacts.
- If greater than the qualify_frequency for an aor, qualify_frequency will be used instead.</synopsis>
- </configOption>
- <configOption name="unidentified_request_period" default="5">
- <synopsis>The number of seconds over which to accumulate unidentified requests.</synopsis>
- <description><para>
- If <literal>unidentified_request_count</literal> unidentified requests are received
- during <literal>unidentified_request_period</literal>, a security event will be generated.
- </para></description>
- </configOption>
- <configOption name="unidentified_request_count" default="5">
- <synopsis>The number of unidentified requests from a single IP to allow.</synopsis>
- <description><para>
- If <literal>unidentified_request_count</literal> unidentified requests are received
- during <literal>unidentified_request_period</literal>, a security event will be generated.
- </para></description>
- </configOption>
- <configOption name="unidentified_request_prune_interval" default="30">
- <synopsis>The interval at which unidentified requests are older than
- twice the unidentified_request_period are pruned.</synopsis>
- </configOption>
- <configOption name="type">
- <synopsis>Must be of type 'global' UNLESS the object name is 'global'.</synopsis>
- </configOption>
- <configOption name="user_agent" default="Asterisk <Asterisk Version>">
- <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
- </configOption>
- <configOption name="regcontext" default="">
- <synopsis>When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given
- peer who registers or unregisters with us.</synopsis>
- </configOption>
- <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
- <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
- </configOption>
- <configOption name="default_voicemail_extension">
- <synopsis>The voicemail extension to send in the NOTIFY Message-Account header if not specified on endpoint or aor</synopsis>
- </configOption>
- <configOption name="debug" default="no">
- <synopsis>Enable/Disable SIP debug logging. Valid options include yes, no, or
- a host address</synopsis>
- </configOption>
- <configOption name="endpoint_identifier_order">
- <synopsis>The order by which endpoint identifiers are processed and checked.
- Identifier names are usually derived from and can be found in the endpoint
- identifier module itself (res_pjsip_endpoint_identifier_*).
- You can use the CLI command "pjsip show identifiers" to see the
- identifiers currently available.</synopsis>
- <description>
- <note><para>
- One of the identifiers is "auth_username" which matches on the username in
- an Authentication header. This method has some security considerations because an
- Authentication header is not present on the first message of a dialog when
- digest authentication is used. The client can't generate it until the server
- sends the challenge in a 401 response. Since Asterisk normally sends a security
- event when an incoming request can't be matched to an endpoint, using auth_username
- requires that the security event be deferred until a request is received with
- the Authentication header and only generated if the username doesn't result in a
- match. This may result in a delay before an attack is recognized. You can control
- how many unmatched requests are received from a single ip address before a security
- event is generated using the unidentified_request parameters.
- </para></note>
- </description>
- </configOption>
- <configOption name="default_from_user" default="asterisk">
- <synopsis>When Asterisk generates an outgoing SIP request, the From header username will be
- set to this value if there is no better option (such as CallerID) to be
- used.</synopsis>
- </configOption>
- <configOption name="default_realm" default="asterisk">
- <synopsis>When Asterisk generates a challenge, the digest realm will be
- set to this value if there is no better option (such as auth/realm) to be
- used.</synopsis>
- </configOption>
- <configOption name="mwi_tps_queue_high" default="500">
- <synopsis>MWI taskprocessor high water alert trigger level.</synopsis>
- <description>
- <para>On a heavily loaded system you may need to adjust the
- taskprocessor queue limits. If any taskprocessor queue size
- reaches its high water level then pjsip will stop processing
- new requests until the alert is cleared. The alert clears
- when all alerting taskprocessor queues have dropped to their
- low water clear level.
- </para>
- </description>
- </configOption>
- <configOption name="mwi_tps_queue_low" default="-1">
- <synopsis>MWI taskprocessor low water clear alert level.</synopsis>
- <description>
- <para>On a heavily loaded system you may need to adjust the
- taskprocessor queue limits. If any taskprocessor queue size
- reaches its high water level then pjsip will stop processing
- new requests until the alert is cleared. The alert clears
- when all alerting taskprocessor queues have dropped to their
- low water clear level.
- </para>
- <note><para>Set to -1 for the low water level to be 90% of
- the high water level.</para></note>
- </description>
- </configOption>
- <configOption name="mwi_disable_initial_unsolicited" default="no">
- <synopsis>Enable/Disable sending unsolicited MWI to all endpoints on startup.</synopsis>
- <description>
- <para>When the initial unsolicited MWI notification are
- enabled on startup then the initial notifications
- get sent at startup. If you have a lot of endpoints
- (thousands) that use unsolicited MWI then you may
- want to consider disabling the initial startup
- notifications.
- </para>
- <para>When the initial unsolicited MWI notifications are
- disabled on startup then the notifications will start
- on the endpoint's next contact update.
- </para>
- </description>
- </configOption>
- <configOption name="ignore_uri_user_options">
- <synopsis>Enable/Disable ignoring SIP URI user field options.</synopsis>
- <description>
- <para>If you have this option enabled and there are semicolons
- in the user field of a SIP URI then the field is truncated
- at the first semicolon. This effectively makes the semicolon
- a non-usable character for PJSIP endpoint names, extensions,
- and AORs. This can be useful for improving compatibility with
- an ITSP that likes to use user options for whatever reason.
- </para>
- <example title="Sample SIP URI">
- sip:1235557890;phone-context=national@x.x.x.x;user=phone
- </example>
- <example title="Sample SIP URI user field">
- 1235557890;phone-context=national
- </example>
- <example title="Sample SIP URI user field truncated">
- 1235557890
- </example>
- <note><para>The caller-id and redirecting number strings
- obtained from incoming SIP URI user fields are always truncated
- at the first semicolon.</para></note>
- </description>
- </configOption>
- <configOption name="use_callerid_contact" default="no">
- <synopsis>Place caller-id information into Contact header</synopsis>
- <description><para>
- This option will cause Asterisk to place caller-id information into
- generated Contact headers.</para>
- </description>
- </configOption>
- <configOption name="send_contact_status_on_update_registration" default="no">
- <synopsis>Enable sending AMI ContactStatus event when a device refreshes its registration.</synopsis>
- </configOption>
- <configOption name="taskprocessor_overload_trigger">
- <synopsis>Trigger scope for taskprocessor overloads</synopsis>
- <description><para>
- This option specifies the trigger the distributor will use for
- detecting taskprocessor overloads. When it detects an overload condition,
- the distrubutor will stop accepting new requests until the overload is
- cleared.
- </para>
- <enumlist>
- <enum name="global"><para>(default) Any taskprocessor overload will trigger.</para></enum>
- <enum name="pjsip_only"><para>Only pjsip taskprocessor overloads will trigger.</para></enum>
- <enum name="none"><para>No overload detection will be performed.</para></enum>
- </enumlist>
- <warning><para>
- The "none" and "pjsip_only" options should be used
- with extreme caution and only to mitigate specific issues.
- Under certain conditions they could make things worse.
- </para></warning>
- </description>
- </configOption>
- <configOption name="norefersub" default="yes">
- <synopsis>Advertise support for RFC4488 REFER subscription suppression</synopsis>
- </configOption>
- </configObject>
- </configFile>
- </configInfo>
- <manager name="PJSIPQualify" language="en_US">
- <synopsis>
- Qualify a chan_pjsip endpoint.
- </synopsis>
- <syntax>
- <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
- <parameter name="Endpoint" required="true">
- <para>The endpoint you want to qualify.</para>
- </parameter>
- </syntax>
- <description>
- <para>Qualify a chan_pjsip endpoint.</para>
- </description>
- </manager>
- <managerEvent language="en_US" name="IdentifyDetail">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about an identify section.</synopsis>
- <syntax>
- <parameter name="ObjectType">
- <para>The object's type. This will always be 'identify'.</para>
- </parameter>
- <parameter name="ObjectName">
- <para>The name of this object.</para>
- </parameter>
- <parameter name="Endpoint">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='endpoint']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SrvLookups">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='srv_lookups']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Match">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='match']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MatchHeader">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='match_header']/synopsis/node())"/></para>
- </parameter>
- <parameter name="EndpointName">
- <para>The name of the endpoint associated with this information.</para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <managerEvent language="en_US" name="AorDetail">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about an Address of Record (AoR) section.</synopsis>
- <syntax>
- <parameter name="ObjectType">
- <para>The object's type. This will always be 'aor'.</para>
- </parameter>
- <parameter name="ObjectName">
- <para>The name of this object.</para>
- </parameter>
- <parameter name="MinimumExpiration">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='minimum_expiration']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MaximumExpiration">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='maximum_expiration']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DefaultExpiration">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='default_expiration']/synopsis/node())"/></para>
- </parameter>
- <parameter name="QualifyFrequency">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='qualify_frequency']/synopsis/node())"/></para>
- </parameter>
- <parameter name="AuthenticateQualify">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='authenticate_qualify']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MaxContacts">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='max_contacts']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RemoveExisting">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='remove_existing']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RemoveUnavailable">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='remove_unavailable']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Mailboxes">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='mailboxes']/synopsis/node())"/></para>
- </parameter>
- <parameter name="OutboundProxy">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='outbound_proxy']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SupportPath">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='support_path']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TotalContacts">
- <para>The total number of contacts associated with this AoR.</para>
- </parameter>
- <parameter name="ContactsRegistered">
- <para>The number of non-permanent contacts associated with this AoR.</para>
- </parameter>
- <parameter name="EndpointName">
- <para>The name of the endpoint associated with this information.</para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <managerEvent language="en_US" name="AuthDetail">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about an authentication section.</synopsis>
- <syntax>
- <parameter name="ObjectType">
- <para>The object's type. This will always be 'auth'.</para>
- </parameter>
- <parameter name="ObjectName">
- <para>The name of this object.</para>
- </parameter>
- <parameter name="Username">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='username']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Password">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='username']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Md5Cred">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='md5_cred']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Realm">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='realm']/synopsis/node())"/></para>
- </parameter>
- <parameter name="NonceLifetime">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='nonce_lifetime']/synopsis/node())"/></para>
- </parameter>
- <parameter name="AuthType">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='auth_type']/synopsis/node())"/></para>
- </parameter>
- <parameter name="EndpointName">
- <para>The name of the endpoint associated with this information.</para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <managerEvent language="en_US" name="TransportDetail">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about an authentication section.</synopsis>
- <syntax>
- <parameter name="ObjectType">
- <para>The object's type. This will always be 'transport'.</para>
- </parameter>
- <parameter name="ObjectName">
- <para>The name of this object.</para>
- </parameter>
- <parameter name="Protocol">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='protocol']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Bind">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='bind']/synopsis/node())"/></para>
- </parameter>
- <parameter name="AsycOperations">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='async_operations']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CaListFile">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='ca_list_file']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CaListPath">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='ca_list_path']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CertFile">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cert_file']/synopsis/node())"/></para>
- </parameter>
- <parameter name="PrivKeyFile">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='priv_key_file']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Password">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='password']/synopsis/node())"/></para>
- </parameter>
- <parameter name="ExternalSignalingAddress">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_signaling_address']/synopsis/node())"/></para>
- </parameter>
- <parameter name="ExternalSignalingPort">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_signaling_port']/synopsis/node())"/></para>
- </parameter>
- <parameter name="ExternalMediaAddress">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_media_address']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Domain">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='domain']/synopsis/node())"/></para>
- </parameter>
- <parameter name="VerifyServer">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='verify_server']/synopsis/node())"/></para>
- </parameter>
- <parameter name="VerifyClient">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='verify_client']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RequireClientCert">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='require_client_cert']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Method">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='method']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Cipher">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cipher']/synopsis/node())"/></para>
- </parameter>
- <parameter name="LocalNet">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='local_net']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Tos">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='tos']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Cos">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cos']/synopsis/node())"/></para>
- </parameter>
- <parameter name="WebsocketWriteTimeout">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='websocket_write_timeout']/synopsis/node())"/></para>
- </parameter>
- <parameter name="EndpointName">
- <para>The name of the endpoint associated with this information.</para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <managerEvent language="en_US" name="EndpointDetail">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about an endpoint section.</synopsis>
- <syntax>
- <parameter name="ObjectType">
- <para>The object's type. This will always be 'endpoint'.</para>
- </parameter>
- <parameter name="ObjectName">
- <para>The name of this object.</para>
- </parameter>
- <parameter name="Context">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='context']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Disallow">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='disallow']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Allow">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtmfMode">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtmf_mode']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RtpIpv6">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_ipv6']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RtpSymmetric">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_symmetric']/synopsis/node())"/></para>
- </parameter>
- <parameter name="IceSupport">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='ice_support']/synopsis/node())"/></para>
- </parameter>
- <parameter name="UsePtime">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='use_ptime']/synopsis/node())"/></para>
- </parameter>
- <parameter name="ForceRport">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='force_rport']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RewriteContact">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rewrite_contact']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Transport">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='transport']/synopsis/node())"/></para>
- </parameter>
- <parameter name="OutboundProxy">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='outbound_proxy']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MohSuggest">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='moh_suggest']/synopsis/node())"/></para>
- </parameter>
- <parameter name="100rel">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='100rel']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Timers">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TimersMinSe">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers_min_se']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TimersSessExpires">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers_sess_expires']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Auth">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='auth']/synopsis/node())"/></para>
- </parameter>
- <parameter name="OutboundAuth">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='outbound_auth']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Aors">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='aors']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MediaAddress">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_address']/synopsis/node())"/></para>
- </parameter>
- <parameter name="IdentifyBy">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='identify_by']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DirectMedia">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DirectMediaMethod">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media_method']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TrustConnectedLine">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_connected_line']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SendConnectedLine">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_connected_line']/synopsis/node())"/></para>
- </parameter>
- <parameter name="ConnectedLineMethod">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='connected_line_method']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DirectMediaGlareMitigation">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media_glare_mitigation']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DisableDirectMediaOnNat">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='disable_direct_media_on_nat']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Callerid">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CalleridPrivacy">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid_privacy']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CalleridTag">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid_tag']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TrustIdInbound">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_id_inbound']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TrustIdOutbound">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_id_outbound']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SendPai">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_pai']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SendRpid">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_rpid']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SendDiversion">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_diversion']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Mailboxes">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='mailboxes']/synopsis/node())"/></para>
- </parameter>
- <parameter name="AggregateMwi">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='aggregate_mwi']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MediaEncryption">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_encryption']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MediaEncryptionOptimistic">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_encryption_optimistic']/synopsis/node())"/></para>
- </parameter>
- <parameter name="UseAvpf">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='use_avpf']/synopsis/node())"/></para>
- </parameter>
- <parameter name="ForceAvp">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='force_avp']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MediaUseReceivedTransport">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_use_received_transport']/synopsis/node())"/></para>
- </parameter>
- <parameter name="OneTouchRecording">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='one_touch_recording']/synopsis/node())"/></para>
- </parameter>
- <parameter name="InbandProgress">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='inband_progress']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CallGroup">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='call_group']/synopsis/node())"/></para>
- </parameter>
- <parameter name="PickupGroup">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='pickup_group']/synopsis/node())"/></para>
- </parameter>
- <parameter name="NamedCallGroup">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='named_call_group']/synopsis/node())"/></para>
- </parameter>
- <parameter name="NamedPickupGroup">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='named_pickup_group']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DeviceStateBusyAt">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='device_state_busy_at']/synopsis/node())"/></para>
- </parameter>
- <parameter name="T38Udptl">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl']/synopsis/node())"/></para>
- </parameter>
- <parameter name="T38UdptlEc">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_ec']/synopsis/node())"/></para>
- </parameter>
- <parameter name="T38UdptlMaxdatagram">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_maxdatagram']/synopsis/node())"/></para>
- </parameter>
- <parameter name="FaxDetect">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='fax_detect']/synopsis/node())"/></para>
- </parameter>
- <parameter name="T38UdptlNat">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_nat']/synopsis/node())"/></para>
- </parameter>
- <parameter name="T38UdptlIpv6">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_ipv6']/synopsis/node())"/></para>
- </parameter>
- <parameter name="T38BindUdptlToMediaAddress">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_bind_udptl_to_media_address']/synopsis/node())"/></para>
- </parameter>
- <parameter name="ToneZone">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tone_zone']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Language">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='language']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RecordOnFeature">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='record_on_feature']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RecordOffFeature">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='record_off_feature']/synopsis/node())"/></para>
- </parameter>
- <parameter name="AllowTransfer">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_transfer']/synopsis/node())"/></para>
- </parameter>
- <parameter name="UserEqPhone">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='user_eq_phone']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MohPassthrough">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='moh_passthrough']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SdpOwner">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_owner']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SdpSession">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_session']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TosAudio">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tos_audio']/synopsis/node())"/></para>
- </parameter>
- <parameter name="TosVideo">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tos_video']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CosAudio">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='cos_audio']/synopsis/node())"/></para>
- </parameter>
- <parameter name="CosVideo">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='cos_video']/synopsis/node())"/></para>
- </parameter>
- <parameter name="AllowSubscribe">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_subscribe']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SubMinExpiry">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sub_min_expiry']/synopsis/node())"/></para>
- </parameter>
- <parameter name="FromUser">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='from_user']/synopsis/node())"/></para>
- </parameter>
- <parameter name="FromDomain">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='from_domain']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MwiFromUser">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='mwi_from_user']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RtpEngine">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_engine']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsVerify">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_verify']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsRekey">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_rekey']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsCertFile">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_cert_file']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsPrivateKey">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_private_key']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsCipher">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_cipher']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsCaFile">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_ca_file']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsCaPath">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_ca_path']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DtlsSetup">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_setup']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SrtpTag32">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='srtp_tag_32']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RedirectMethod">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='redirect_method']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SetVar">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='set_var']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MessageContext">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='message_context']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Accountcode">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='accountcode']/synopsis/node())"/></para>
- </parameter>
- <parameter name="PreferredCodecOnly">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='preferred_codec_only']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DeviceState">
- <para>The aggregate device state for this endpoint.</para>
- </parameter>
- <parameter name="ActiveChannels">
- <para>The number of active channels associated with this endpoint.</para>
- </parameter>
- <parameter name="SubscribeContext">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='subscribe_context']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Allowoverlap">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_overlap']/synopsis/node())"/></para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <managerEvent language="en_US" name="AorList">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about an Address of Record (AoR) section.</synopsis>
- <syntax>
- <parameter name="ObjectType">
- <para>The object's type. This will always be 'aor'.</para>
- </parameter>
- <parameter name="ObjectName">
- <para>The name of this object.</para>
- </parameter>
- <parameter name="MinimumExpiration">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='minimum_expiration']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MaximumExpiration">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='maximum_expiration']/synopsis/node())"/></para>
- </parameter>
- <parameter name="DefaultExpiration">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='default_expiration']/synopsis/node())"/></para>
- </parameter>
- <parameter name="QualifyFrequency">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='qualify_frequency']/synopsis/node())"/></para>
- </parameter>
- <parameter name="AuthenticateQualify">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='authenticate_qualify']/synopsis/node())"/></para>
- </parameter>
- <parameter name="MaxContacts">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='max_contacts']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RemoveExisting">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='remove_existing']/synopsis/node())"/></para>
- </parameter>
- <parameter name="RemoveUnavailable">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='remove_unavailable']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Mailboxes">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='mailboxes']/synopsis/node())"/></para>
- </parameter>
- <parameter name="OutboundProxy">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='outbound_proxy']/synopsis/node())"/></para>
- </parameter>
- <parameter name="SupportPath">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='support_path']/synopsis/node())"/></para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <managerEvent language="en_US" name="AuthList">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about an Address of Record (Auth) section.</synopsis>
- <syntax>
- <parameter name="ObjectType">
- <para>The object's type. This will always be 'auth'.</para>
- </parameter>
- <parameter name="ObjectName">
- <para>The name of this object.</para>
- </parameter>
- <parameter name="Username">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='username']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Md5Cred">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='md5_cred']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Realm">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='realm']/synopsis/node())"/></para>
- </parameter>
- <parameter name="AuthType">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='auth_type']/synopsis/node())"/></para>
- </parameter>
- <parameter name="Password">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='password']/synopsis/node())"/></para>
- </parameter>
- <parameter name="NonceLifetime">
- <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='nonce_lifetime']/synopsis/node())"/></para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <managerEvent language="en_US" name="ContactList">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about a contact section.</synopsis>
- <syntax>
- <parameter name="ObjectType">
- <para>The object's type. This will always be 'contact'.</para>
- </parameter>
- <parameter name="ObjectName">
- <para>The name of this object.</para>
- </parameter>
- <parameter name="ViaAddr">
- <para>IP address of the last Via header in REGISTER request.
- Will only appear in the event if available.</para>
- </parameter>
- <parameter name="ViaPort">
- <para>Port number of the last Via header in REGISTER request.
- Will only appear in the event if available.</para>
- </parameter>
- <parameter name="QualifyTimeout">
- <para>The elapsed time in decimal seconds after which an OPTIONS
- message is sent before the contact is considered unavailable.</para>
- </parameter>
- <parameter name="CallId">
- <para>Content of the Call-ID header in REGISTER request.
- Will only appear in the event if available.</para>
- </parameter>
- <parameter name="RegServer">
- <para>Asterisk Server name.</para>
- </parameter>
- <parameter name="PruneOnBoot">
- <para>If true delete the contact on Asterisk restart/boot.</para>
- </parameter>
- <parameter name="Path">
- <para>The Path header received on the REGISTER.</para>
- </parameter>
- <parameter name="Endpoint">
- <para>The name of the endpoint associated with this information.</para>
- </parameter>
- <parameter name="AuthenticateQualify">
- <para>A boolean indicating whether a qualify should be authenticated.</para>
- </parameter>
- <parameter name="Uri">
- <para>This contact's URI.</para>
- </parameter>
- <parameter name="QualifyFrequency">
- <para>The interval in seconds at which the contact will be qualified.</para>
- </parameter>
- <parameter name="UserAgent">
- <para>Content of the User-Agent header in REGISTER request</para>
- </parameter>
- <parameter name="ExpirationTime">
- <para>Absolute time that this contact is no longer valid after</para>
- </parameter>
- <parameter name="OutboundProxy">
- <para>The contact's outbound proxy.</para>
- </parameter>
- <parameter name="Status">
- <para>This contact's status.</para>
- <enumlist>
- <enum name="Reachable"/>
- <enum name="Unreachable"/>
- <enum name="NonQualified"/>
- <enum name="Unknown"/>
- </enumlist>
- </parameter>
- <parameter name="RoundtripUsec">
- <para>The round trip time in microseconds.</para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <managerEvent language="en_US" name="ContactStatusDetail">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about a contact's status.</synopsis>
- <syntax>
- <parameter name="AOR">
- <para>The AoR that owns this contact.</para>
- </parameter>
- <parameter name="URI">
- <para>This contact's URI.</para>
- </parameter>
- <parameter name="Status">
- <para>This contact's status.</para>
- <enumlist>
- <enum name="Reachable"/>
- <enum name="Unreachable"/>
- <enum name="NonQualified"/>
- <enum name="Unknown"/>
- </enumlist>
- </parameter>
- <parameter name="RoundtripUsec">
- <para>The round trip time in microseconds.</para>
- </parameter>
- <parameter name="EndpointName">
- <para>The name of the endpoint associated with this information.</para>
- </parameter>
- <parameter name="UserAgent">
- <para>Content of the User-Agent header in REGISTER request</para>
- </parameter>
- <parameter name="RegExpire">
- <para>Absolute time that this contact is no longer valid after</para>
- </parameter>
- <parameter name="ViaAddress">
- <para>IP address:port of the last Via header in REGISTER request.
- Will only appear in the event if available.</para>
- </parameter>
- <parameter name="CallID">
- <para>Content of the Call-ID header in REGISTER request.
- Will only appear in the event if available.</para>
- </parameter>
- <parameter name="ID">
- <para>The sorcery ID of the contact.</para>
- </parameter>
- <parameter name="AuthenticateQualify">
- <para>A boolean indicating whether a qualify should be authenticated.</para>
- </parameter>
- <parameter name="OutboundProxy">
- <para>The contact's outbound proxy.</para>
- </parameter>
- <parameter name="Path">
- <para>The Path header received on the REGISTER.</para>
- </parameter>
- <parameter name="QualifyFrequency">
- <para>The interval in seconds at which the contact will be qualified.</para>
- </parameter>
- <parameter name="QualifyTimeout">
- <para>The elapsed time in decimal seconds after which an OPTIONS
- message is sent before the contact is considered unavailable.</para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <managerEvent language="en_US" name="EndpointList">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide details about a contact's status.</synopsis>
- <syntax>
- <parameter name="ObjectType">
- <para>The object's type. This will always be 'endpoint'.</para>
- </parameter>
- <parameter name="ObjectName">
- <para>The name of this object.</para>
- </parameter>
- <parameter name="Transport">
- <para>The transport configurations associated with this endpoint.</para>
- </parameter>
- <parameter name="Aor">
- <para>The aor configurations associated with this endpoint.</para>
- </parameter>
- <parameter name="Auths">
- <para>The inbound authentication configurations associated with this endpoint.</para>
- </parameter>
- <parameter name="OutboundAuths">
- <para>The outbound authentication configurations associated with this endpoint.</para>
- </parameter>
- <parameter name="DeviceState">
- <para>The aggregate device state for this endpoint.</para>
- </parameter>
- <parameter name="ActiveChannels">
- <para>The number of active channels associated with this endpoint.</para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- <manager name="PJSIPShowEndpoints" language="en_US">
- <synopsis>
- Lists PJSIP endpoints.
- </synopsis>
- <syntax />
- <description>
- <para>
- Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
- is raised that contains relevant attributes and status information. Once all
- endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
- </para>
- </description>
- <responses>
- <list-elements>
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='EndpointList'])" />
- </list-elements>
- <managerEvent language="en_US" name="EndpointListComplete">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide final information about an endpoint list.</synopsis>
- <syntax>
- <parameter name="EventList"/>
- <parameter name="ListItems"/>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- </responses>
- </manager>
- <manager name="PJSIPShowEndpoint" language="en_US">
- <synopsis>
- Detail listing of an endpoint and its objects.
- </synopsis>
- <syntax>
- <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
- <parameter name="Endpoint" required="true">
- <para>The endpoint to list.</para>
- </parameter>
- </syntax>
- <description>
- <para>
- Provides a detailed listing of options for a given endpoint. Events are issued
- showing the configuration and status of the endpoint and associated objects. These
- events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
- <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
- <literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
- associated (for instance AoRs). Once all detail events have been raised a final
- <literal>EndpointDetailComplete</literal> event is issued.
- </para>
- </description>
- <responses>
- <list-elements>
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='EndpointDetail'])" />
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='IdentifyDetail'])" />
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='ContactStatusDetail'])" />
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='AuthDetail'])" />
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='TransportDetail'])" />
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='AorDetail'])" />
- </list-elements>
- <managerEvent language="en_US" name="EndpointDetailComplete">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide final information about endpoint details.</synopsis>
- <syntax>
- <parameter name="EventList"/>
- <parameter name="ListItems"/>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- </responses>
- </manager>
- <manager name="PJSIPShowAors" language="en_US">
- <synopsis>
- Lists PJSIP AORs.
- </synopsis>
- <syntax />
- <description>
- <para>
- Provides a listing of all AORs. For each AOR an <literal>AorList</literal> event
- is raised that contains relevant attributes and status information. Once all
- aors have been listed an <literal>AorListComplete</literal> event is issued.
- </para>
- </description>
- <responses>
- <list-elements>
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='AorList'])" />
- </list-elements>
- <managerEvent language="en_US" name="AorListComplete">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide final information about an aor list.</synopsis>
- <syntax>
- <parameter name="EventList"/>
- <parameter name="ListItems"/>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- </responses>
- </manager>
- <manager name="PJSIPShowAuths" language="en_US">
- <synopsis>
- Lists PJSIP Auths.
- </synopsis>
- <syntax />
- <description>
- <para>Provides a listing of all Auths. For each Auth an <literal>AuthList</literal> event
- is raised that contains relevant attributes and status information. Once all
- auths have been listed an <literal>AuthListComplete</literal> event is issued.
- </para>
- </description>
- <responses>
- <list-elements>
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='AuthList'])" />
- </list-elements>
- <managerEvent language="en_US" name="AuthListComplete">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide final information about an auth list.</synopsis>
- <syntax>
- <parameter name="EventList"/>
- <parameter name="ListItems"/>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- </responses>
- </manager>
- <manager name="PJSIPShowContacts" language="en_US">
- <synopsis>
- Lists PJSIP Contacts.
- </synopsis>
- <syntax />
- <description>
- <para>Provides a listing of all Contacts. For each Contact a <literal>ContactList</literal>
- event is raised that contains relevant attributes and status information.
- Once all contacts have been listed a <literal>ContactListComplete</literal> event
- is issued.
- </para>
- </description>
- <responses>
- <list-elements>
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='ContactList'])" />
- </list-elements>
- <managerEvent language="en_US" name="ContactListComplete">
- <managerEventInstance class="EVENT_FLAG_COMMAND">
- <synopsis>Provide final information about a contact list.</synopsis>
- <syntax>
- <parameter name="EventList"/>
- <parameter name="ListItems"/>
- </syntax>
- </managerEventInstance>
- </managerEvent>
- </responses>
- </manager>
-
- ***/
-
#define MOD_DATA_CONTACT "contact"
/*! Number of serializers in pool if one not supplied. */
--- /dev/null
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE docs SYSTEM "appdocsxml.dtd">
+<docs>
+ <configInfo name="res_pjsip" language="en_US">
+ <synopsis>SIP Resource using PJProject</synopsis>
+ <configFile name="pjsip.conf">
+ <configObject name="endpoint">
+ <synopsis>Endpoint</synopsis>
+ <description><para>
+ The <emphasis>Endpoint</emphasis> is the primary configuration object.
+ It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
+ dialable entries of their own. Communication with another SIP device is
+ accomplished via Addresses of Record (AoRs) which have one or more
+ contacts associated with them. Endpoints <emphasis>NOT</emphasis> configured to
+ use a <literal>transport</literal> will default to first transport found
+ in <filename>pjsip.conf</filename> that matches its type.
+ </para>
+ <para>Example: An Endpoint has been configured with no transport.
+ When it comes time to call an AoR, PJSIP will find the
+ first transport that matches the type. A SIP URI of <literal>sip:5000@[11::33]</literal>
+ will use the first IPv6 transport and try to send the request.
+ </para>
+ <para>If the anonymous endpoint identifier is in use an endpoint with the name
+ "anonymous@domain" will be searched for as a last resort. If this is not found
+ it will fall back to searching for "anonymous". If neither endpoints are found
+ the anonymous endpoint identifier will not return an endpoint and anonymous
+ calling will not be possible.
+ </para>
+ </description>
+ <configOption name="100rel" default="yes">
+ <synopsis>Allow support for RFC3262 provisional ACK tags</synopsis>
+ <description>
+ <enumlist>
+ <enum name="no" />
+ <enum name="required" />
+ <enum name="yes" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="aggregate_mwi" default="yes">
+ <synopsis>Condense MWI notifications into a single NOTIFY.</synopsis>
+ <description><para>When enabled, <replaceable>aggregate_mwi</replaceable> condenses message
+ waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled,
+ individual NOTIFYs are sent for each mailbox.</para></description>
+ </configOption>
+ <configOption name="allow">
+ <synopsis>Media Codec(s) to allow</synopsis>
+ </configOption>
+ <configOption name="allow_overlap" default="yes">
+ <synopsis>Enable RFC3578 overlap dialing support.</synopsis>
+ </configOption>
+ <configOption name="aors">
+ <synopsis>AoR(s) to be used with the endpoint</synopsis>
+ <description><para>
+ List of comma separated AoRs that the endpoint should be associated with.
+ </para></description>
+ </configOption>
+ <configOption name="auth">
+ <synopsis>Authentication Object(s) associated with the endpoint</synopsis>
+ <description><para>
+ This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
+ in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
+ </para><para>
+ Endpoints without an authentication object
+ configured will allow connections without verification.</para>
+ <note><para>
+ Using the same auth section for inbound and outbound
+ authentication is not recommended. There is a difference in
+ meaning for an empty realm setting between inbound and outbound
+ authentication uses. See the auth realm description for details.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="callerid">
+ <synopsis>CallerID information for the endpoint</synopsis>
+ <description><para>
+ Must be in the format <literal>Name <Number></literal>,
+ or only <literal><Number></literal>.
+ </para></description>
+ </configOption>
+ <configOption name="callerid_privacy">
+ <synopsis>Default privacy level</synopsis>
+ <description>
+ <enumlist>
+ <enum name="allowed_not_screened" />
+ <enum name="allowed_passed_screen" />
+ <enum name="allowed_failed_screen" />
+ <enum name="allowed" />
+ <enum name="prohib_not_screened" />
+ <enum name="prohib_passed_screen" />
+ <enum name="prohib_failed_screen" />
+ <enum name="prohib" />
+ <enum name="unavailable" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="callerid_tag">
+ <synopsis>Internal id_tag for the endpoint</synopsis>
+ </configOption>
+ <configOption name="context">
+ <synopsis>Dialplan context for inbound sessions</synopsis>
+ </configOption>
+ <configOption name="direct_media_glare_mitigation" default="none">
+ <synopsis>Mitigation of direct media (re)INVITE glare</synopsis>
+ <description>
+ <para>
+ This setting attempts to avoid creating INVITE glare scenarios
+ by disabling direct media reINVITEs in one direction thereby allowing
+ designated servers (according to this option) to initiate direct
+ media reINVITEs without contention and significantly reducing call
+ setup time.
+ </para>
+ <para>
+ A more detailed description of how this option functions can be found on
+ the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
+ </para>
+ <enumlist>
+ <enum name="none" />
+ <enum name="outgoing" />
+ <enum name="incoming" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="direct_media_method" default="invite">
+ <synopsis>Direct Media method type</synopsis>
+ <description>
+ <para>Method for setting up Direct Media between endpoints.</para>
+ <enumlist>
+ <enum name="invite" />
+ <enum name="reinvite">
+ <para>Alias for the <literal>invite</literal> value.</para>
+ </enum>
+ <enum name="update" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="trust_connected_line">
+ <synopsis>Accept Connected Line updates from this endpoint</synopsis>
+ </configOption>
+ <configOption name="send_connected_line">
+ <synopsis>Send Connected Line updates to this endpoint</synopsis>
+ </configOption>
+ <configOption name="connected_line_method" default="invite">
+ <synopsis>Connected line method type</synopsis>
+ <description>
+ <para>Method used when updating connected line information.</para>
+ <enumlist>
+ <enum name="invite">
+ <para>When set to <literal>invite</literal>, check the remote's Allow header and
+ if UPDATE is allowed, send UPDATE instead of INVITE to avoid SDP
+ renegotiation. If UPDATE is not Allowed, send INVITE.</para>
+ </enum>
+ <enum name="reinvite">
+ <para>Alias for the <literal>invite</literal> value.</para>
+ </enum>
+ <enum name="update">
+ <para>If set to <literal>update</literal>, send UPDATE regardless of what the remote
+ Allows. </para>
+ </enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="direct_media" default="yes">
+ <synopsis>Determines whether media may flow directly between endpoints.</synopsis>
+ </configOption>
+ <configOption name="disable_direct_media_on_nat" default="no">
+ <synopsis>Disable direct media session refreshes when NAT obstructs the media session</synopsis>
+ </configOption>
+ <configOption name="disallow">
+ <synopsis>Media Codec(s) to disallow</synopsis>
+ </configOption>
+ <configOption name="dtmf_mode" default="rfc4733">
+ <synopsis>DTMF mode</synopsis>
+ <description>
+ <para>This setting allows to choose the DTMF mode for endpoint communication.</para>
+ <enumlist>
+ <enum name="rfc4733">
+ <para>DTMF is sent out of band of the main audio stream. This
+ supercedes the older <emphasis>RFC-2833</emphasis> used within
+ the older <literal>chan_sip</literal>.</para>
+ </enum>
+ <enum name="inband">
+ <para>DTMF is sent as part of audio stream.</para>
+ </enum>
+ <enum name="info">
+ <para>DTMF is sent as SIP INFO packets.</para>
+ </enum>
+ <enum name="auto">
+ <para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</para>
+ </enum>
+ <enum name="auto_info">
+ <para>DTMF is sent as RFC 4733 if the other side supports it or as SIP INFO if not.</para>
+ </enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="media_address">
+ <synopsis>IP address used in SDP for media handling</synopsis>
+ <description><para>
+ At the time of SDP creation, the IP address defined here will be used as
+ the media address for individual streams in the SDP.
+ </para>
+ <note><para>
+ Be aware that the <literal>external_media_address</literal> option, set in Transport
+ configuration, can also affect the final media address used in the SDP.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="bind_rtp_to_media_address">
+ <synopsis>Bind the RTP instance to the media_address</synopsis>
+ <description><para>
+ If media_address is specified, this option causes the RTP instance to be bound to the
+ specified ip address which causes the packets to be sent from that address.
+ </para>
+ </description>
+ </configOption>
+ <configOption name="force_rport" default="yes">
+ <synopsis>Force use of return port</synopsis>
+ </configOption>
+ <configOption name="ice_support" default="no">
+ <synopsis>Enable the ICE mechanism to help traverse NAT</synopsis>
+ </configOption>
+ <configOption name="identify_by">
+ <synopsis>Way(s) for the endpoint to be identified</synopsis>
+ <description>
+ <para>Endpoints and AORs can be identified in multiple ways. This
+ option is a comma separated list of methods the endpoint can be
+ identified.
+ </para>
+ <note><para>
+ This option controls both how an endpoint is matched for incoming
+ traffic and also how an AOR is determined if a registration
+ occurs. You must list at least one method that also matches for
+ AORs or the registration will fail.
+ </para></note>
+ <enumlist>
+ <enum name="username">
+ <para>Matches the endpoint or AOR ID based on the username
+ and domain in the From header (or To header for AORs). If
+ an exact match on both username and domain/realm fails, the
+ match is retried with just the username.
+ </para>
+ </enum>
+ <enum name="auth_username">
+ <para>Matches the endpoint or AOR ID based on the username
+ and realm in the Authentication header. If an exact match
+ on both username and domain/realm fails, the match is
+ retried with just the username.
+ </para>
+ <note><para>This method of identification has some security
+ considerations because an Authentication header is not
+ present on the first message of a dialog when digest
+ authentication is used. The client can't generate it until
+ the server sends the challenge in a 401 response. Since
+ Asterisk normally sends a security event when an incoming
+ request can't be matched to an endpoint, using this method
+ requires that the security event be deferred until a request
+ is received with the Authentication header and only
+ generated if the username doesn't result in a match. This
+ may result in a delay before an attack is recognized. You
+ can control how many unmatched requests are received from
+ a single ip address before a security event is generated
+ using the <literal>unidentified_request</literal>
+ parameters in the "global" configuration object.
+ </para></note>
+ </enum>
+ <enum name="ip">
+ <para>Matches the endpoint based on the source IP address.
+ </para>
+ <para>This method of identification is not configured here
+ but simply allowed by this configuration option. See the
+ documentation for the <literal>identify</literal>
+ configuration section for more details on this method of
+ endpoint identification.
+ </para>
+ </enum>
+ <enum name="header">
+ <para>Matches the endpoint based on a configured SIP header
+ value.
+ </para>
+ <para>This method of identification is not configured here
+ but simply allowed by this configuration option. See the
+ documentation for the <literal>identify</literal>
+ configuration section for more details on this method of
+ endpoint identification.
+ </para>
+ </enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="redirect_method">
+ <synopsis>How redirects received from an endpoint are handled</synopsis>
+ <description><para>
+ When a redirect is received from an endpoint there are multiple ways it can be handled.
+ If this option is set to <literal>user</literal> the user portion of the redirect target
+ is treated as an extension within the dialplan and dialed using a Local channel. If this option
+ is set to <literal>uri_core</literal> the target URI is returned to the dialing application
+ which dials it using the PJSIP channel driver and endpoint originally used. If this option is
+ set to <literal>uri_pjsip</literal> the redirect occurs within chan_pjsip itself and is not exposed
+ to the core at all. The <literal>uri_pjsip</literal> option has the benefit of being more efficient
+ and also supporting multiple potential redirect targets. The con is that since redirection occurs
+ within chan_pjsip redirecting information is not forwarded and redirection can not be
+ prevented.
+ </para>
+ <enumlist>
+ <enum name="user" />
+ <enum name="uri_core" />
+ <enum name="uri_pjsip" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="mailboxes">
+ <synopsis>NOTIFY the endpoint when state changes for any of the specified mailboxes</synopsis>
+ <description><para>
+ Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state
+ changes happen for any of the specified mailboxes. More than one mailbox can be
+ specified with a comma-delimited string. app_voicemail mailboxes must be specified
+ as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by
+ external sources, such as through the res_mwi_external module, you must specify
+ strings supported by the external system.
+ </para><para>
+ For endpoints that SUBSCRIBE for MWI, use the <literal>mailboxes</literal> option in your AOR
+ configuration.
+ </para></description>
+ </configOption>
+ <configOption name="mwi_subscribe_replaces_unsolicited">
+ <synopsis>An MWI subscribe will replace sending unsolicited NOTIFYs</synopsis>
+ </configOption>
+ <configOption name="voicemail_extension">
+ <synopsis>The voicemail extension to send in the NOTIFY Message-Account header</synopsis>
+ </configOption>
+ <configOption name="moh_suggest" default="default">
+ <synopsis>Default Music On Hold class</synopsis>
+ </configOption>
+ <configOption name="outbound_auth">
+ <synopsis>Authentication object(s) used for outbound requests</synopsis>
+ <description><para>
+ This is a comma-delimited list of <replaceable>auth</replaceable>
+ sections defined in <filename>pjsip.conf</filename> used to respond
+ to outbound connection authentication challenges.</para>
+ <note><para>
+ Using the same auth section for inbound and outbound
+ authentication is not recommended. There is a difference in
+ meaning for an empty realm setting between inbound and outbound
+ authentication uses. See the auth realm description for details.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="outbound_proxy">
+ <synopsis>Full SIP URI of the outbound proxy used to send requests</synopsis>
+ </configOption>
+ <configOption name="rewrite_contact">
+ <synopsis>Allow Contact header to be rewritten with the source IP address-port</synopsis>
+ <description><para>
+ On inbound SIP messages from this endpoint, the Contact header or an
+ appropriate Record-Route header will be changed to have the source IP
+ address and port. This option does not affect outbound messages sent to
+ this endpoint. This option helps servers communicate with endpoints
+ that are behind NATs. This option also helps reuse reliable transport
+ connections such as TCP and TLS.
+ </para></description>
+ </configOption>
+ <configOption name="rtp_ipv6" default="no">
+ <synopsis>Allow use of IPv6 for RTP traffic</synopsis>
+ </configOption>
+ <configOption name="rtp_symmetric" default="no">
+ <synopsis>Enforce that RTP must be symmetric</synopsis>
+ </configOption>
+ <configOption name="send_diversion" default="yes">
+ <synopsis>Send the Diversion header, conveying the diversion
+ information to the called user agent</synopsis>
+ </configOption>
+ <configOption name="send_history_info" default="no">
+ <synopsis>Send the History-Info header, conveying the diversion
+ information to the called and calling user agents</synopsis>
+ </configOption>
+ <configOption name="send_pai" default="no">
+ <synopsis>Send the P-Asserted-Identity header</synopsis>
+ </configOption>
+ <configOption name="send_rpid" default="no">
+ <synopsis>Send the Remote-Party-ID header</synopsis>
+ </configOption>
+ <configOption name="rpid_immediate" default="no">
+ <synopsis>Immediately send connected line updates on unanswered incoming calls.</synopsis>
+ <description>
+ <para>When enabled, immediately send <emphasis>180 Ringing</emphasis>
+ or <emphasis>183 Progress</emphasis> response messages to the
+ caller if the connected line information is updated before
+ the call is answered. This can send a <emphasis>180 Ringing</emphasis>
+ response before the call has even reached the far end. The
+ caller can start hearing ringback before the far end even gets
+ the call. Many phones tend to grab the first connected line
+ information and refuse to update the display if it changes. The
+ first information is not likely to be correct if the call
+ goes to an endpoint not under the control of this Asterisk
+ box.</para>
+ <para>When disabled, a connected line update must wait for
+ another reason to send a message with the connected line
+ information to the caller before the call is answered. You can
+ trigger the sending of the information by using an appropriate
+ dialplan application such as <emphasis>Ringing</emphasis>.</para>
+ </description>
+ </configOption>
+ <configOption name="timers_min_se" default="90">
+ <synopsis>Minimum session timers expiration period</synopsis>
+ <description><para>
+ Minimum session timer expiration period. Time in seconds.
+ </para></description>
+ </configOption>
+ <configOption name="timers" default="yes">
+ <synopsis>Session timers for SIP packets</synopsis>
+ <description>
+ <enumlist>
+ <enum name="no" />
+ <enum name="yes" />
+ <enum name="required" />
+ <enum name="always" />
+ <enum name="forced"><para>Alias of always</para></enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="timers_sess_expires" default="1800">
+ <synopsis>Maximum session timer expiration period</synopsis>
+ <description><para>
+ Maximum session timer expiration period. Time in seconds.
+ </para></description>
+ </configOption>
+ <configOption name="transport">
+ <synopsis>Explicit transport configuration to use</synopsis>
+ <description>
+ <para>This will <emphasis>force</emphasis> the endpoint to use the
+ specified transport configuration to send SIP messages. You need
+ to already know what kind of transport (UDP/TCP/IPv4/etc) the
+ endpoint device will use.
+ </para>
+ <note><para>Not specifying a transport will select the first
+ configured transport in <filename>pjsip.conf</filename> which is
+ compatible with the URI we are trying to contact.
+ </para></note>
+ <warning><para>Transport configuration is not affected by reloads. In order to
+ change transports, a full Asterisk restart is required</para></warning>
+ </description>
+ </configOption>
+ <configOption name="trust_id_inbound" default="no">
+ <synopsis>Accept identification information received from this endpoint</synopsis>
+ <description><para>This option determines whether Asterisk will accept
+ identification from the endpoint from headers such as P-Asserted-Identity
+ or Remote-Party-ID header. This option applies both to calls originating from the
+ endpoint and calls originating from Asterisk. If <literal>no</literal>, the
+ configured Caller-ID from pjsip.conf will always be used as the identity for
+ the endpoint.</para></description>
+ </configOption>
+ <configOption name="trust_id_outbound" default="no">
+ <synopsis>Send private identification details to the endpoint.</synopsis>
+ <description><para>This option determines whether res_pjsip will send private
+ identification information to the endpoint. If <literal>no</literal>,
+ private Caller-ID information will not be forwarded to the endpoint.
+ "Private" in this case refers to any method of restricting identification.
+ Example: setting <replaceable>callerid_privacy</replaceable> to any
+ <literal>prohib</literal> variation.
+ Example: If <replaceable>trust_id_inbound</replaceable> is set to
+ <literal>yes</literal>, the presence of a <literal>Privacy: id</literal>
+ header in a SIP request or response would indicate the identification
+ provided in the request is private.</para></description>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be of type 'endpoint'.</synopsis>
+ </configOption>
+ <configOption name="use_ptime" default="no">
+ <synopsis>Use Endpoint's requested packetization interval</synopsis>
+ </configOption>
+ <configOption name="use_avpf" default="no">
+ <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
+ endpoint.</synopsis>
+ <description><para>
+ If set to <literal>yes</literal>, res_pjsip will use the AVPF or SAVPF RTP
+ profile for all media offers on outbound calls and media updates and will
+ decline media offers not using the AVPF or SAVPF profile.
+ </para><para>
+ If set to <literal>no</literal>, res_pjsip will use the AVP or SAVP RTP
+ profile for all media offers on outbound calls and media updates, and will
+ decline media offers not using the AVP or SAVP profile.
+ </para></description>
+ </configOption>
+ <configOption name="force_avp" default="no">
+ <synopsis>Determines whether res_pjsip will use and enforce usage of AVP,
+ regardless of the RTP profile in use for this endpoint.</synopsis>
+ <description><para>
+ If set to <literal>yes</literal>, res_pjsip will use the AVP, AVPF, SAVP, or
+ SAVPF RTP profile for all media offers on outbound calls and media updates including
+ those for DTLS-SRTP streams.
+ </para><para>
+ If set to <literal>no</literal>, res_pjsip will use the respective RTP profile
+ depending on configuration.
+ </para></description>
+ </configOption>
+ <configOption name="media_use_received_transport" default="no">
+ <synopsis>Determines whether res_pjsip will use the media transport received in the
+ offer SDP in the corresponding answer SDP.</synopsis>
+ <description><para>
+ If set to <literal>yes</literal>, res_pjsip will use the received media transport.
+ </para><para>
+ If set to <literal>no</literal>, res_pjsip will use the respective RTP profile
+ depending on configuration.
+ </para></description>
+ </configOption>
+ <configOption name="media_encryption" default="no">
+ <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
+ for this endpoint.</synopsis>
+ <description>
+ <enumlist>
+ <enum name="no"><para>
+ res_pjsip will offer no encryption and allow no encryption to be setup.
+ </para></enum>
+ <enum name="sdes"><para>
+ res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
+ transport should be used in conjunction with this option to prevent
+ exposure of media encryption keys.
+ </para></enum>
+ <enum name="dtls"><para>
+ res_pjsip will offer DTLS-SRTP setup.
+ </para></enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="media_encryption_optimistic" default="no">
+ <synopsis>Determines whether encryption should be used if possible but does not terminate the
+ session if not achieved.</synopsis>
+ <description><para>
+ This option only applies if <replaceable>media_encryption</replaceable> is
+ set to <literal>sdes</literal> or <literal>dtls</literal>.
+ </para></description>
+ </configOption>
+ <configOption name="g726_non_standard" default="no">
+ <synopsis>Force g.726 to use AAL2 packing order when negotiating g.726 audio</synopsis>
+ <description><para>
+ When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
+ packing order instead of what is recommended by RFC3551. Since this essentially
+ replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
+ specified in the endpoint's allowed codec list.
+ </para></description>
+ </configOption>
+ <configOption name="inband_progress" default="no">
+ <synopsis>Determines whether chan_pjsip will indicate ringing using inband
+ progress.</synopsis>
+ <description><para>
+ If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
+ when told to indicate ringing and will immediately start sending ringing
+ as audio.
+ </para><para>
+ If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
+ to indicate ringing and will NOT send it as audio.
+ </para></description>
+ </configOption>
+ <configOption name="call_group">
+ <synopsis>The numeric pickup groups for a channel.</synopsis>
+ <description><para>
+ Can be set to a comma separated list of numbers or ranges between the values
+ of 0-63 (maximum of 64 groups).
+ </para></description>
+ </configOption>
+ <configOption name="pickup_group">
+ <synopsis>The numeric pickup groups that a channel can pickup.</synopsis>
+ <description><para>
+ Can be set to a comma separated list of numbers or ranges between the values
+ of 0-63 (maximum of 64 groups).
+ </para></description>
+ </configOption>
+ <configOption name="named_call_group">
+ <synopsis>The named pickup groups for a channel.</synopsis>
+ <description><para>
+ Can be set to a comma separated list of case sensitive strings limited by
+ supported line length.
+ </para></description>
+ </configOption>
+ <configOption name="named_pickup_group">
+ <synopsis>The named pickup groups that a channel can pickup.</synopsis>
+ <description><para>
+ Can be set to a comma separated list of case sensitive strings limited by
+ supported line length.
+ </para></description>
+ </configOption>
+ <configOption name="device_state_busy_at" default="0">
+ <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
+ <description><para>
+ When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
+ PJSIP channel driver will return busy as the device state instead of in use.
+ </para></description>
+ </configOption>
+ <configOption name="t38_udptl" default="no">
+ <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis>
+ <description><para>
+ If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted
+ and relayed.
+ </para></description>
+ </configOption>
+ <configOption name="t38_udptl_ec" default="none">
+ <synopsis>T.38 UDPTL error correction method</synopsis>
+ <description>
+ <enumlist>
+ <enum name="none"><para>
+ No error correction should be used.
+ </para></enum>
+ <enum name="fec"><para>
+ Forward error correction should be used.
+ </para></enum>
+ <enum name="redundancy"><para>
+ Redundancy error correction should be used.
+ </para></enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="t38_udptl_maxdatagram" default="0">
+ <synopsis>T.38 UDPTL maximum datagram size</synopsis>
+ <description><para>
+ This option can be set to override the maximum datagram of a remote endpoint for broken
+ endpoints.
+ </para></description>
+ </configOption>
+ <configOption name="fax_detect" default="no">
+ <synopsis>Whether CNG tone detection is enabled</synopsis>
+ <description><para>
+ This option can be set to send the session to the fax extension when a CNG tone is
+ detected.
+ </para></description>
+ </configOption>
+ <configOption name="fax_detect_timeout">
+ <synopsis>How long into a call before fax_detect is disabled for the call</synopsis>
+ <description><para>
+ The option determines how many seconds into a call before the
+ fax_detect option is disabled for the call. Setting the value
+ to zero disables the timeout.
+ </para></description>
+ </configOption>
+ <configOption name="t38_udptl_nat" default="no">
+ <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis>
+ <description><para>
+ When enabled the UDPTL stack will send UDPTL packets to the source address of
+ received packets.
+ </para></description>
+ </configOption>
+ <configOption name="t38_udptl_ipv6" default="no">
+ <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis>
+ <description><para>
+ When enabled the UDPTL stack will use IPv6.
+ </para></description>
+ </configOption>
+ <configOption name="t38_bind_udptl_to_media_address" default="no">
+ <synopsis>Bind the UDPTL instance to the media_adress</synopsis>
+ <description><para>
+ If media_address is specified, this option causes the UDPTL instance to be bound to
+ the specified ip address which causes the packets to be sent from that address.
+ </para></description>
+ </configOption>
+ <configOption name="tone_zone">
+ <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis>
+ </configOption>
+ <configOption name="language">
+ <synopsis>Set the default language to use for channels created for this endpoint.</synopsis>
+ </configOption>
+ <configOption name="one_touch_recording" default="no">
+ <synopsis>Determines whether one-touch recording is allowed for this endpoint.</synopsis>
+ <see-also>
+ <ref type="configOption">record_on_feature</ref>
+ <ref type="configOption">record_off_feature</ref>
+ </see-also>
+ </configOption>
+ <configOption name="record_on_feature" default="automixmon">
+ <synopsis>The feature to enact when one-touch recording is turned on.</synopsis>
+ <description>
+ <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this
+ feature will be enabled for the channel. The feature designated here can be any built-in
+ or dynamic feature defined in features.conf.</para>
+ <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
+ </description>
+ <see-also>
+ <ref type="configOption">one_touch_recording</ref>
+ <ref type="configOption">record_off_feature</ref>
+ </see-also>
+ </configOption>
+ <configOption name="record_off_feature" default="automixmon">
+ <synopsis>The feature to enact when one-touch recording is turned off.</synopsis>
+ <description>
+ <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this
+ feature will be enabled for the channel. The feature designated here can be any built-in
+ or dynamic feature defined in features.conf.</para>
+ <note><para>This setting has no effect if the endpoint's one_touch_recording option is disabled</para></note>
+ </description>
+ <see-also>
+ <ref type="configOption">one_touch_recording</ref>
+ <ref type="configOption">record_on_feature</ref>
+ </see-also>
+ </configOption>
+ <configOption name="rtp_engine" default="asterisk">
+ <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis>
+ </configOption>
+ <configOption name="allow_transfer" default="yes">
+ <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis>
+ </configOption>
+ <configOption name="user_eq_phone" default="no">
+ <synopsis>Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number</synopsis>
+ </configOption>
+ <configOption name="moh_passthrough" default="no">
+ <synopsis>Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side</synopsis>
+ </configOption>
+ <configOption name="sdp_owner" default="-">
+ <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
+ </configOption>
+ <configOption name="sdp_session" default="Asterisk">
+ <synopsis>String used for the SDP session (s=) line.</synopsis>
+ </configOption>
+ <configOption name="tos_audio">
+ <synopsis>DSCP TOS bits for audio streams</synopsis>
+ <description><para>
+ See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
+ </para></description>
+ </configOption>
+ <configOption name="tos_video">
+ <synopsis>DSCP TOS bits for video streams</synopsis>
+ <description><para>
+ See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
+ </para></description>
+ </configOption>
+ <configOption name="cos_audio">
+ <synopsis>Priority for audio streams</synopsis>
+ <description><para>
+ See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
+ </para></description>
+ </configOption>
+ <configOption name="cos_video">
+ <synopsis>Priority for video streams</synopsis>
+ <description><para>
+ See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
+ </para></description>
+ </configOption>
+ <configOption name="allow_subscribe" default="yes">
+ <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
+ </configOption>
+ <configOption name="sub_min_expiry" default="60">
+ <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis>
+ </configOption>
+ <configOption name="from_user">
+ <synopsis>Username to use in From header for requests to this endpoint.</synopsis>
+ </configOption>
+ <configOption name="mwi_from_user">
+ <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis>
+ </configOption>
+ <configOption name="from_domain">
+ <synopsis>Domain to user in From header for requests to this endpoint.</synopsis>
+ </configOption>
+ <configOption name="dtls_verify">
+ <synopsis>Verify that the provided peer certificate is valid</synopsis>
+ <description><para>
+ This option only applies if <replaceable>media_encryption</replaceable> is
+ set to <literal>dtls</literal>.
+ </para><para>
+ It can be one of the following values:
+ </para><enumlist>
+ <enum name="no"><para>
+ meaning no verification is done.
+ </para></enum>
+ <enum name="fingerprint"><para>
+ meaning to verify the remote fingerprint.
+ </para></enum>
+ <enum name="certificate"><para>
+ meaning to verify the remote certificate.
+ </para></enum>
+ <enum name="yes"><para>
+ meaning to verify both the remote fingerprint and certificate.
+ </para></enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="dtls_rekey">
+ <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis>
+ <description><para>
+ This option only applies if <replaceable>media_encryption</replaceable> is
+ set to <literal>dtls</literal>.
+ </para><para>
+ If this is not set or the value provided is 0 rekeying will be disabled.
+ </para></description>
+ </configOption>
+ <configOption name="dtls_auto_generate_cert" default="no">
+ <synopsis>Whether or not to automatically generate an ephemeral X.509 certificate</synopsis>
+ <description>
+ <para>
+ If enabled, Asterisk will generate an X.509 certificate for each DTLS session.
+ This option only applies if <replaceable>media_encryption</replaceable> is set
+ to <literal>dtls</literal>. This option will be automatically enabled if
+ <literal>webrtc</literal> is enabled and <literal>dtls_cert_file</literal> is
+ not specified.
+ </para>
+ </description>
+ </configOption>
+ <configOption name="dtls_cert_file">
+ <synopsis>Path to certificate file to present to peer</synopsis>
+ <description><para>
+ This option only applies if <replaceable>media_encryption</replaceable> is
+ set to <literal>dtls</literal>.
+ </para></description>
+ </configOption>
+ <configOption name="dtls_private_key">
+ <synopsis>Path to private key for certificate file</synopsis>
+ <description><para>
+ This option only applies if <replaceable>media_encryption</replaceable> is
+ set to <literal>dtls</literal>.
+ </para></description>
+ </configOption>
+ <configOption name="dtls_cipher">
+ <synopsis>Cipher to use for DTLS negotiation</synopsis>
+ <description><para>
+ This option only applies if <replaceable>media_encryption</replaceable> is
+ set to <literal>dtls</literal>.
+ </para>
+ <para>Many options for acceptable ciphers. See link for more:</para>
+ <para>http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
+ </para></description>
+ </configOption>
+ <configOption name="dtls_ca_file">
+ <synopsis>Path to certificate authority certificate</synopsis>
+ <description><para>
+ This option only applies if <replaceable>media_encryption</replaceable> is
+ set to <literal>dtls</literal>.
+ </para></description>
+ </configOption>
+ <configOption name="dtls_ca_path">
+ <synopsis>Path to a directory containing certificate authority certificates</synopsis>
+ <description><para>
+ This option only applies if <replaceable>media_encryption</replaceable> is
+ set to <literal>dtls</literal>.
+ </para></description>
+ </configOption>
+ <configOption name="dtls_setup">
+ <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis>
+ <description>
+ <para>
+ This option only applies if <replaceable>media_encryption</replaceable> is
+ set to <literal>dtls</literal>.
+ </para>
+ <enumlist>
+ <enum name="active"><para>
+ res_pjsip will make a connection to the peer.
+ </para></enum>
+ <enum name="passive"><para>
+ res_pjsip will accept connections from the peer.
+ </para></enum>
+ <enum name="actpass"><para>
+ res_pjsip will offer and accept connections from the peer.
+ </para></enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="dtls_fingerprint">
+ <synopsis>Type of hash to use for the DTLS fingerprint in the SDP.</synopsis>
+ <description>
+ <para>
+ This option only applies if <replaceable>media_encryption</replaceable> is
+ set to <literal>dtls</literal>.
+ </para>
+ <enumlist>
+ <enum name="SHA-256"></enum>
+ <enum name="SHA-1"></enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="srtp_tag_32">
+ <synopsis>Determines whether 32 byte tags should be used instead of 80 byte tags.</synopsis>
+ <description><para>
+ This option only applies if <replaceable>media_encryption</replaceable> is
+ set to <literal>sdes</literal> or <literal>dtls</literal>.
+ </para></description>
+ </configOption>
+ <configOption name="set_var">
+ <synopsis>Variable set on a channel involving the endpoint.</synopsis>
+ <description><para>
+ When a new channel is created using the endpoint set the specified
+ variable(s) on that channel. For multiple channel variables specify
+ multiple 'set_var'(s).
+ </para></description>
+ </configOption>
+ <configOption name="message_context">
+ <synopsis>Context to route incoming MESSAGE requests to.</synopsis>
+ <description><para>
+ If specified, incoming MESSAGE requests will be routed to the indicated
+ dialplan context. If no <replaceable>message_context</replaceable> is
+ specified, then the <replaceable>context</replaceable> setting is used.
+ </para></description>
+ </configOption>
+ <configOption name="accountcode">
+ <synopsis>An accountcode to set automatically on any channels created for this endpoint.</synopsis>
+ <description><para>
+ If specified, any channel created for this endpoint will automatically
+ have this accountcode set on it.
+ </para></description>
+ </configOption>
+ <configOption name="preferred_codec_only" default="no">
+ <synopsis>Respond to a SIP invite with the single most preferred codec rather than advertising all joint codec capabilities. This limits the other side's codec choice to exactly what we prefer.</synopsis>
+ </configOption>
+ <configOption name="rtp_keepalive">
+ <synopsis>Number of seconds between RTP comfort noise keepalive packets.</synopsis>
+ <description><para>
+ At the specified interval, Asterisk will send an RTP comfort noise frame. This may
+ be useful for situations where Asterisk is behind a NAT or firewall and must keep
+ a hole open in order to allow for media to arrive at Asterisk.
+ </para></description>
+ </configOption>
+ <configOption name="rtp_timeout" default="0">
+ <synopsis>Maximum number of seconds without receiving RTP (while off hold) before terminating call.</synopsis>
+ <description><para>
+ This option configures the number of seconds without RTP (while off hold) before
+ considering a channel as dead. When the number of seconds is reached the underlying
+ channel is hung up. By default this option is set to 0, which means do not check.
+ </para></description>
+ </configOption>
+ <configOption name="rtp_timeout_hold" default="0">
+ <synopsis>Maximum number of seconds without receiving RTP (while on hold) before terminating call.</synopsis>
+ <description><para>
+ This option configures the number of seconds without RTP (while on hold) before
+ considering a channel as dead. When the number of seconds is reached the underlying
+ channel is hung up. By default this option is set to 0, which means do not check.
+ </para></description>
+ </configOption>
+ <configOption name="acl">
+ <synopsis>List of IP ACL section names in acl.conf</synopsis>
+ <description><para>
+ This matches sections configured in <literal>acl.conf</literal>. The value is
+ defined as a list of comma-delimited section names.
+ </para></description>
+ </configOption>
+ <configOption name="deny">
+ <synopsis>List of IP addresses to deny access from</synopsis>
+ <description><para>
+ The value is a comma-delimited list of IP addresses. IP addresses may
+ have a subnet mask appended. The subnet mask may be written in either
+ CIDR or dotted-decimal notation. Separate the IP address and subnet
+ mask with a slash ('/')
+ </para></description>
+ </configOption>
+ <configOption name="permit">
+ <synopsis>List of IP addresses to permit access from</synopsis>
+ <description><para>
+ The value is a comma-delimited list of IP addresses. IP addresses may
+ have a subnet mask appended. The subnet mask may be written in either
+ CIDR or dotted-decimal notation. Separate the IP address and subnet
+ mask with a slash ('/')
+ </para></description>
+ </configOption>
+ <configOption name="contact_acl">
+ <synopsis>List of Contact ACL section names in acl.conf</synopsis>
+ <description><para>
+ This matches sections configured in <literal>acl.conf</literal>. The value is
+ defined as a list of comma-delimited section names.
+ </para></description>
+ </configOption>
+ <configOption name="contact_deny">
+ <synopsis>List of Contact header addresses to deny</synopsis>
+ <description><para>
+ The value is a comma-delimited list of IP addresses. IP addresses may
+ have a subnet mask appended. The subnet mask may be written in either
+ CIDR or dotted-decimal notation. Separate the IP address and subnet
+ mask with a slash ('/')
+ </para></description>
+ </configOption>
+ <configOption name="contact_permit">
+ <synopsis>List of Contact header addresses to permit</synopsis>
+ <description><para>
+ The value is a comma-delimited list of IP addresses. IP addresses may
+ have a subnet mask appended. The subnet mask may be written in either
+ CIDR or dotted-decimal notation. Separate the IP address and subnet
+ mask with a slash ('/')
+ </para></description>
+ </configOption>
+ <configOption name="subscribe_context">
+ <synopsis>Context for incoming MESSAGE requests.</synopsis>
+ <description><para>
+ If specified, incoming SUBSCRIBE requests will be searched for the matching
+ extension in the indicated context.
+ If no <replaceable>subscribe_context</replaceable> is specified,
+ then the <replaceable>context</replaceable> setting is used.
+ </para></description>
+ </configOption>
+ <configOption name="contact_user" default="">
+ <synopsis>Force the user on the outgoing Contact header to this value.</synopsis>
+ <description><para>
+ On outbound requests, force the user portion of the Contact header to this value.
+ </para></description>
+ </configOption>
+ <configOption name="asymmetric_rtp_codec" default="no">
+ <synopsis>Allow the sending and receiving RTP codec to differ</synopsis>
+ <description><para>
+ When set to "yes" the codec in use for sending will be allowed to differ from
+ that of the received one. PJSIP will not automatically switch the sending one
+ to the receiving one.
+ </para></description>
+ </configOption>
+ <configOption name="rtcp_mux" default="no">
+ <synopsis>Enable RFC 5761 RTCP multiplexing on the RTP port</synopsis>
+ <description><para>
+ With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux"
+ attribute on all media streams. This will result in RTP and RTCP being sent and received
+ on the same port. This shifts the demultiplexing logic to the application rather than
+ the transport layer. This option is useful when interoperating with WebRTC endpoints
+ since they mandate this option's use.
+ </para></description>
+ </configOption>
+ <configOption name="refer_blind_progress" default="yes">
+ <synopsis>Whether to notifies all the progress details on blind transfer</synopsis>
+ <description><para>
+ Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK"
+ after REFER has been accepted. If set to <literal>no</literal> then asterisk
+ will not send the progress details, but immediately will send "200 OK".
+ </para></description>
+ </configOption>
+ <configOption name="notify_early_inuse_ringing" default="no">
+ <synopsis>Whether to notifies dialog-info 'early' on InUse&Ringing state</synopsis>
+ <description><para>
+ Control whether dialog-info subscriptions get 'early' state
+ on Ringing when already INUSE.
+ </para></description>
+ </configOption>
+ <configOption name="max_audio_streams" default="1">
+ <synopsis>The maximum number of allowed audio streams for the endpoint</synopsis>
+ <description><para>
+ This option enforces a limit on the maximum simultaneous negotiated audio
+ streams allowed for the endpoint.
+ </para></description>
+ </configOption>
+ <configOption name="max_video_streams" default="1">
+ <synopsis>The maximum number of allowed video streams for the endpoint</synopsis>
+ <description><para>
+ This option enforces a limit on the maximum simultaneous negotiated video
+ streams allowed for the endpoint.
+ </para></description>
+ </configOption>
+ <configOption name="bundle" default="no">
+ <synopsis>Enable RTP bundling</synopsis>
+ <description><para>
+ With this option enabled, Asterisk will attempt to negotiate the use of bundle.
+ If negotiated this will result in multiple RTP streams being carried over the same
+ underlying transport. Note that enabling bundle will also enable the rtcp_mux option.
+ </para></description>
+ </configOption>
+ <configOption name="webrtc" default="no">
+ <synopsis>Defaults and enables some options that are relevant to WebRTC</synopsis>
+ <description><para>
+ When set to "yes" this also enables the following values that are needed in
+ order for basic WebRTC support to work: rtcp_mux, use_avpf, ice_support, and
+ use_received_transport. The following configuration settings also get defaulted
+ as follows:</para>
+ <para>media_encryption=dtls</para>
+ <para>dtls_auto_generate_cert=yes (if dtls_cert_file is not set)</para>
+ <para>dtls_verify=fingerprint</para>
+ <para>dtls_setup=actpass</para>
+ </description>
+ </configOption>
+ <configOption name="incoming_mwi_mailbox">
+ <synopsis>Mailbox name to use when incoming MWI NOTIFYs are received</synopsis>
+ <description><para>
+ If an MWI NOTIFY is received <emphasis>from</emphasis> this endpoint,
+ this mailbox will be used when notifying other modules of MWI status
+ changes. If not set, incoming MWI NOTIFYs are ignored.
+ </para></description>
+ </configOption>
+ <configOption name="follow_early_media_fork">
+ <synopsis>Follow SDP forked media when To tag is different</synopsis>
+ <description><para>
+ On outgoing calls, if the UAS responds with different SDP attributes
+ on subsequent 18X or 2XX responses (such as a port update) AND the
+ To tag on the subsequent response is different than that on the previous
+ one, follow it. This usually happens when the INVITE is forked to multiple
+ UASs and more than one sends an SDP answer.
+ </para>
+ <note><para>
+ This option must also be enabled in the <literal>system</literal>
+ section for it to take effect here.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="accept_multiple_sdp_answers" default="no">
+ <synopsis>Accept multiple SDP answers on non-100rel responses</synopsis>
+ <description><para>
+ On outgoing calls, if the UAS responds with different SDP attributes
+ on non-100rel 18X or 2XX responses (such as a port update) AND the
+ To tag on the subsequent response is the same as that on the previous one,
+ process the updated SDP. This can happen when the UAS needs to change ports
+ for some reason such as using a separate port for custom ringback.
+ </para>
+ <note><para>
+ This option must also be enabled in the <literal>system</literal>
+ section for it to take effect here.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="suppress_q850_reason_headers" default="no">
+ <synopsis>Suppress Q.850 Reason headers for this endpoint</synopsis>
+ <description><para>
+ Some devices can't accept multiple Reason headers and get confused
+ when both 'SIP' and 'Q.850' Reason headers are received. This
+ option allows the 'Q.850' Reason header to be suppressed.</para>
+ </description>
+ </configOption>
+ <configOption name="ignore_183_without_sdp" default="no">
+ <synopsis>Do not forward 183 when it doesn't contain SDP</synopsis>
+ <description><para>
+ Certain SS7 internetworking scenarios can result in a 183
+ to be generated for reasons other than early media. Forwarding
+ this 183 can cause loss of ringback tone. This flag emulates
+ the behavior of chan_sip and prevents these 183 responses from
+ being forwarded.</para>
+ </description>
+ </configOption>
+ <configOption name="stir_shaken" default="no">
+ <synopsis>Enable STIR/SHAKEN support on this endpoint</synopsis>
+ <description><para>
+ Enable STIR/SHAKEN support on this endpoint. On incoming INVITEs,
+ the Identity header will be checked for validity. On outgoing
+ INVITEs, an Identity header will be added.</para>
+ </description>
+ </configOption>
+ <configOption name="allow_unauthenticated_options" default="no">
+ <synopsis>Skip authentication when receiving OPTIONS requests</synopsis>
+ <description><para>
+ RFC 3261 says that the response to an OPTIONS request MUST be the
+ same had the request been an INVITE. Some UAs use OPTIONS requests
+ like a 'ping' and the expectation is that they will return a
+ 200 OK.</para>
+ <para>Enabling <literal>allow_unauthenticated_options</literal>
+ will skip authentication of OPTIONS requests for the given
+ endpoint.</para>
+ <para>There are security implications to enabling this setting as
+ it can allow information disclosure to occur - specifically, if
+ enabled, an external party could enumerate and find the endpoint
+ name by sending OPTIONS requests and examining the
+ responses.</para>
+ </description>
+ </configOption>
+ </configObject>
+ <configObject name="auth">
+ <synopsis>Authentication type</synopsis>
+ <description><para>
+ Authentication objects hold the authentication information for use
+ by other objects such as <literal>endpoints</literal> or <literal>registrations</literal>.
+ This also allows for multiple objects to use a single auth object. See
+ the <literal>auth_type</literal> config option for password style choices.
+ </para></description>
+ <configOption name="auth_type" default="userpass">
+ <synopsis>Authentication type</synopsis>
+ <description><para>
+ This option specifies which of the password style config options should be read
+ when trying to authenticate an endpoint inbound request. If set to <literal>userpass</literal>
+ then we'll read from the 'password' option. For <literal>md5</literal> we'll read
+ from 'md5_cred'. The following values are valid:
+ </para>
+ <enumlist>
+ <enum name="md5"/>
+ <enum name="userpass"/>
+ </enumlist>
+ <para>
+ </para>
+ <note>
+ <para>
+ This setting only describes whether the password is in
+ plain text or has been pre-hashed with MD5. It doesn't describe
+ the acceptable digest algorithms we'll accept in a received
+ challenge.
+ </para>
+ </note>
+ </description>
+ </configOption>
+ <configOption name="nonce_lifetime" default="32">
+ <synopsis>Lifetime of a nonce associated with this authentication config.</synopsis>
+ </configOption>
+ <configOption name="md5_cred" default="">
+ <synopsis>MD5 Hash used for authentication.</synopsis>
+ <description><para>
+ Only used when auth_type is <literal>md5</literal>.
+ As an alternative to specifying a plain text password,
+ you can hash the username, realm and password
+ together one time and place the hash value here.
+ The input to the hash function must be in the
+ following format:
+ </para>
+ <para>
+ </para>
+ <para>
+ <username>:<realm>:<password>
+ </para>
+ <para>
+ </para>
+ <para>
+ For incoming authentication (asterisk is the server),
+ the realm must match either the realm set in this object
+ or the <variable>default_realm</variable> set in in the
+ <replaceable>global</replaceable> object.
+ </para>
+ <para>
+ </para>
+ <para>
+ For outgoing authentication (asterisk is the UAC),
+ the realm must match what the server will be sending
+ in their WWW-Authenticate header. It can't be blank
+ unless you expect the server to be sending a blank
+ realm in the header. You can't use pre-hashed
+ passwords with a wildcard auth object.
+ You can generate the hash with the following shell
+ command:
+ </para>
+ <para>
+ </para>
+ <para>
+ $ echo -n "myname:myrealm:mypassword" | md5sum
+ </para>
+ <para>
+ </para>
+ <para>
+ Note the '-n'. You don't want a newline to be part
+ of the hash.
+ </para></description>
+ </configOption>
+ <configOption name="password">
+ <synopsis>Plain text password used for authentication.</synopsis>
+ <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
+ </configOption>
+ <configOption name="realm" default="">
+ <synopsis>SIP realm for endpoint</synopsis>
+ <description><para>
+ For incoming authentication (asterisk is the UAS),
+ this is the realm to be sent on WWW-Authenticate
+ headers. If not specified, the <replaceable>global</replaceable>
+ object's <variable>default_realm</variable> will be used.
+ </para>
+ <para>
+ </para>
+ <para>
+ For outgoing authentication (asterisk is the UAC), this
+ must either be the realm the server is expected to send,
+ or left blank or contain a single '*' to automatically
+ use the realm sent by the server. If you have multiple
+ auth objects for an endpoint, the realm is also used to
+ match the auth object to the realm the server sent.
+ </para>
+ <para>
+ </para>
+ <note>
+ <para>
+ Using the same auth section for inbound and outbound
+ authentication is not recommended. There is a difference in
+ meaning for an empty realm setting between inbound and outbound
+ authentication uses.
+ </para>
+ </note>
+ <para>
+ </para>
+ <note>
+ <para>
+ If more than one auth object with the same realm or
+ more than one wildcard auth object associated to
+ an endpoint, we can only use the first one of
+ each defined on the endpoint.
+ </para>
+ </note>
+ </description>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be 'auth'</synopsis>
+ </configOption>
+ <configOption name="username">
+ <synopsis>Username to use for account</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="domain_alias">
+ <synopsis>Domain Alias</synopsis>
+ <description><para>
+ Signifies that a domain is an alias. If the domain on a session is
+ not found to match an AoR then this object is used to see if we have
+ an alias for the AoR to which the endpoint is binding. This objects
+ name as defined in configuration should be the domain alias and a
+ config option is provided to specify the domain to be aliased.
+ </para></description>
+ <configOption name="type">
+ <synopsis>Must be of type 'domain_alias'.</synopsis>
+ </configOption>
+ <configOption name="domain">
+ <synopsis>Domain to be aliased</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="transport">
+ <synopsis>SIP Transport</synopsis>
+ <description><para>
+ <emphasis>Transports</emphasis>
+ </para>
+ <para>There are different transports and protocol derivatives
+ supported by <literal>res_pjsip</literal>. They are in order of
+ preference: UDP, TCP, and WebSocket (WS).</para>
+ <note><para>Changes to transport configuration in pjsip.conf will only be
+ effected on a complete restart of Asterisk. A module reload
+ will not suffice.</para></note>
+ </description>
+ <configOption name="async_operations" default="1">
+ <synopsis>Number of simultaneous Asynchronous Operations</synopsis>
+ </configOption>
+ <configOption name="bind">
+ <synopsis>IP Address and optional port to bind to for this transport</synopsis>
+ </configOption>
+ <configOption name="ca_list_file">
+ <synopsis>File containing a list of certificates to read (TLS ONLY, not WSS)</synopsis>
+ </configOption>
+ <configOption name="ca_list_path">
+ <synopsis>Path to directory containing a list of certificates to read (TLS ONLY, not WSS)</synopsis>
+ </configOption>
+ <configOption name="cert_file">
+ <synopsis>Certificate file for endpoint (TLS ONLY, not WSS)</synopsis>
+ <description><para>
+ A path to a .crt or .pem file can be provided. However, only
+ the certificate is read from the file, not the private key.
+ The <literal>priv_key_file</literal> option must supply a
+ matching key file.
+ </para></description>
+ </configOption>
+ <configOption name="cipher">
+ <synopsis>Preferred cryptography cipher names (TLS ONLY, not WSS)</synopsis>
+ <description>
+ <para>Comma separated list of cipher names or numeric equivalents.
+ Numeric equivalents can be either decimal or hexadecimal (0xX).
+ </para>
+ <para>There are many cipher names. Use the CLI command
+ <literal>pjsip list ciphers</literal> to see a list of cipher
+ names available for your installation. See link for more:</para>
+ <para>http://www.openssl.org/docs/apps/ciphers.html#CIPHER_SUITE_NAMES
+ </para>
+ </description>
+ </configOption>
+ <configOption name="domain">
+ <synopsis>Domain the transport comes from</synopsis>
+ </configOption>
+ <configOption name="external_media_address">
+ <synopsis>External IP address to use in RTP handling</synopsis>
+ <description><para>
+ When a request or response is sent out, if the destination of the
+ message is outside the IP network defined in the option <literal>localnet</literal>,
+ and the media address in the SDP is within the localnet network, then the
+ media address in the SDP will be rewritten to the value defined for
+ <literal>external_media_address</literal>.
+ </para></description>
+ </configOption>
+ <configOption name="external_signaling_address">
+ <synopsis>External address for SIP signalling</synopsis>
+ </configOption>
+ <configOption name="external_signaling_port" default="0">
+ <synopsis>External port for SIP signalling</synopsis>
+ </configOption>
+ <configOption name="method">
+ <synopsis>Method of SSL transport (TLS ONLY, not WSS)</synopsis>
+ <description>
+ <enumlist>
+ <enum name="default">
+ <para>The default as defined by PJSIP. This is currently TLSv1, but may change with future releases.</para>
+ </enum>
+ <enum name="unspecified">
+ <para>This option is equivalent to setting 'default'</para>
+ </enum>
+ <enum name="tlsv1" />
+ <enum name="tlsv1_1" />
+ <enum name="tlsv1_2" />
+ <enum name="sslv2" />
+ <enum name="sslv3" />
+ <enum name="sslv23" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="local_net">
+ <synopsis>Network to consider local (used for NAT purposes).</synopsis>
+ <description><para>This must be in CIDR or dotted decimal format with the IP
+ and mask separated with a slash ('/').</para></description>
+ </configOption>
+ <configOption name="password">
+ <synopsis>Password required for transport</synopsis>
+ </configOption>
+ <configOption name="priv_key_file">
+ <synopsis>Private key file (TLS ONLY, not WSS)</synopsis>
+ </configOption>
+ <configOption name="protocol" default="udp">
+ <synopsis>Protocol to use for SIP traffic</synopsis>
+ <description>
+ <enumlist>
+ <enum name="udp" />
+ <enum name="tcp" />
+ <enum name="tls" />
+ <enum name="ws" />
+ <enum name="wss" />
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="require_client_cert" default="false">
+ <synopsis>Require client certificate (TLS ONLY, not WSS)</synopsis>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be of type 'transport'.</synopsis>
+ </configOption>
+ <configOption name="verify_client" default="false">
+ <synopsis>Require verification of client certificate (TLS ONLY, not WSS)</synopsis>
+ </configOption>
+ <configOption name="verify_server" default="false">
+ <synopsis>Require verification of server certificate (TLS ONLY, not WSS)</synopsis>
+ </configOption>
+ <configOption name="tos" default="false">
+ <synopsis>Enable TOS for the signalling sent over this transport</synopsis>
+ <description>
+ <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
+ for more information on this parameter.</para>
+ <note><para>This option does not apply to the <replaceable>ws</replaceable>
+ or the <replaceable>wss</replaceable> protocols.</para></note>
+ </description>
+ </configOption>
+ <configOption name="cos" default="false">
+ <synopsis>Enable COS for the signalling sent over this transport</synopsis>
+ <description>
+ <para>See <literal>https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service</literal>
+ for more information on this parameter.</para>
+ <note><para>This option does not apply to the <replaceable>ws</replaceable>
+ or the <replaceable>wss</replaceable> protocols.</para></note>
+ </description>
+ </configOption>
+ <configOption name="websocket_write_timeout" default="100">
+ <synopsis>The timeout (in milliseconds) to set on WebSocket connections.</synopsis>
+ <description>
+ <para>If a websocket connection accepts input slowly, the timeout
+ for writes to it can be increased to keep it from being disconnected.
+ Value is in milliseconds.</para>
+ </description>
+ </configOption>
+ <configOption name="allow_reload" default="no">
+ <synopsis>Allow this transport to be reloaded.</synopsis>
+ <description>
+ <para>Allow this transport to be reloaded when res_pjsip is reloaded.
+ This option defaults to "no" because reloading a transport may disrupt
+ in-progress calls.</para>
+ </description>
+ </configOption>
+ <configOption name="symmetric_transport" default="no">
+ <synopsis>Use the same transport for outgoing requests as incoming ones.</synopsis>
+ <description>
+ <para>When a request from a dynamic contact
+ comes in on a transport with this option set to 'yes',
+ the transport name will be saved and used for subsequent
+ outgoing requests like OPTIONS, NOTIFY and INVITE. It's
+ saved as a contact uri parameter named 'x-ast-txp' and will
+ display with the contact uri in CLI, AMI, and ARI output.
+ On the outgoing request, if a transport wasn't explicitly
+ set on the endpoint AND the request URI is not a hostname,
+ the saved transport will be used and the 'x-ast-txp'
+ parameter stripped from the outgoing packet.
+ </para>
+ </description>
+ </configOption>
+ </configObject>
+ <configObject name="contact">
+ <synopsis>A way of creating an aliased name to a SIP URI</synopsis>
+ <description><para>
+ Contacts are a way to hide SIP URIs from the dialplan directly.
+ They are also used to make a group of contactable parties when
+ in use with <literal>AoR</literal> lists.
+ </para></description>
+ <configOption name="type">
+ <synopsis>Must be of type 'contact'.</synopsis>
+ </configOption>
+ <configOption name="uri">
+ <synopsis>SIP URI to contact peer</synopsis>
+ </configOption>
+ <configOption name="expiration_time">
+ <synopsis>Time to keep alive a contact</synopsis>
+ <description><para>
+ Time to keep alive a contact. String style specification.
+ </para></description>
+ </configOption>
+ <configOption name="qualify_frequency" default="0">
+ <synopsis>Interval at which to qualify a contact</synopsis>
+ <description><para>
+ Interval between attempts to qualify the contact for reachability.
+ If <literal>0</literal> never qualify. Time in seconds.
+ </para></description>
+ </configOption>
+ <configOption name="qualify_timeout" default="3.0">
+ <synopsis>Timeout for qualify</synopsis>
+ <description><para>
+ If the contact doesn't respond to the OPTIONS request before the timeout,
+ the contact is marked unavailable.
+ If <literal>0</literal> no timeout. Time in fractional seconds.
+ </para></description>
+ </configOption>
+ <configOption name="authenticate_qualify">
+ <synopsis>Authenticates a qualify challenge response if needed</synopsis>
+ <description>
+ <para>If true and a qualify request receives a challenge response then
+ authentication is attempted before declaring the contact available.
+ </para>
+ <note><para>This option does nothing as we will always complete
+ the challenge response authentication if the qualify request is
+ challenged.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="outbound_proxy">
+ <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
+ <description><para>
+ If set the provided URI will be used as the outbound proxy when an
+ OPTIONS request is sent to a contact for qualify purposes.
+ </para></description>
+ </configOption>
+ <configOption name="path">
+ <synopsis>Stored Path vector for use in Route headers on outgoing requests.</synopsis>
+ </configOption>
+ <configOption name="user_agent">
+ <synopsis>User-Agent header from registration.</synopsis>
+ <description><para>
+ The User-Agent is automatically stored based on data present in incoming SIP
+ REGISTER requests and is not intended to be configured manually.
+ </para></description>
+ </configOption>
+ <configOption name="endpoint">
+ <synopsis>Endpoint name</synopsis>
+ <description><para>
+ The name of the endpoint this contact belongs to
+ </para></description>
+ </configOption>
+ <configOption name="reg_server">
+ <synopsis>Asterisk Server name</synopsis>
+ <description><para>
+ Asterisk Server name on which SIP endpoint registered.
+ </para></description>
+ </configOption>
+ <configOption name="via_addr">
+ <synopsis>IP-address of the last Via header from registration.</synopsis>
+ <description><para>
+ The last Via header should contain the address of UA which sent the request.
+ The IP-address of the last Via header is automatically stored based on data present
+ in incoming SIP REGISTER requests and is not intended to be configured manually.
+ </para></description>
+ </configOption>
+ <configOption name="via_port">
+ <synopsis>IP-port of the last Via header from registration.</synopsis>
+ <description><para>
+ The IP-port of the last Via header is automatically stored based on data present
+ in incoming SIP REGISTER requests and is not intended to be configured manually.
+ </para></description>
+ </configOption>
+ <configOption name="call_id">
+ <synopsis>Call-ID header from registration.</synopsis>
+ <description><para>
+ The Call-ID header is automatically stored based on data present
+ in incoming SIP REGISTER requests and is not intended to be configured manually.
+ </para></description>
+ </configOption>
+ <configOption name="prune_on_boot">
+ <synopsis>A contact that cannot survive a restart/boot.</synopsis>
+ <description><para>
+ The option is set if the incoming SIP REGISTER contact is rewritten
+ on a reliable transport and is not intended to be configured manually.
+ </para></description>
+ </configOption>
+ </configObject>
+ <configObject name="aor">
+ <synopsis>The configuration for a location of an endpoint</synopsis>
+ <description><para>
+ An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
+ AoRs are specified, an endpoint will not be reachable by Asterisk.
+ Beyond that, an AoR has other uses within Asterisk, such as inbound
+ registration.
+ </para><para>
+ An <literal>AoR</literal> is a way to allow dialing a group
+ of <literal>Contacts</literal> that all use the same
+ <literal>endpoint</literal> for calls.
+ </para><para>
+ This can be used as another way of grouping a list of contacts to dial
+ rather than specifying them each directly when dialing via the dialplan.
+ This must be used in conjunction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
+ </para><para>
+ Registrations: For Asterisk to match an inbound registration to an endpoint,
+ the AoR object name must match the user portion of the SIP URI in the "To:"
+ header of the inbound SIP registration. That will usually be equivalent
+ to the "user name" set in your hard or soft phones configuration.
+ </para></description>
+ <configOption name="contact">
+ <synopsis>Permanent contacts assigned to AoR</synopsis>
+ <description><para>
+ Contacts specified will be called whenever referenced
+ by <literal>chan_pjsip</literal>.
+ </para><para>
+ Use a separate "contact=" entry for each contact required. Contacts
+ are specified using a SIP URI.
+ </para></description>
+ </configOption>
+ <configOption name="default_expiration" default="3600">
+ <synopsis>Default expiration time in seconds for contacts that are dynamically bound to an AoR.</synopsis>
+ </configOption>
+ <configOption name="mailboxes">
+ <synopsis>Allow subscriptions for the specified mailbox(es)</synopsis>
+ <description><para>This option applies when an external entity subscribes to an AoR
+ for Message Waiting Indications. The mailboxes specified will be subscribed to.
+ More than one mailbox can be specified with a comma-delimited string.
+ app_voicemail mailboxes must be specified as mailbox@context;
+ for example: mailboxes=6001@default. For mailboxes provided by external sources,
+ such as through the res_mwi_external module, you must specify strings supported by
+ the external system.
+ </para><para>
+ For endpoints that cannot SUBSCRIBE for MWI, you can set the <literal>mailboxes</literal> option in your
+ endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint.
+ </para></description>
+ </configOption>
+ <configOption name="voicemail_extension">
+ <synopsis>The voicemail extension to send in the NOTIFY Message-Account header</synopsis>
+ </configOption>
+ <configOption name="maximum_expiration" default="7200">
+ <synopsis>Maximum time to keep an AoR</synopsis>
+ <description><para>
+ Maximum time to keep a peer with explicit expiration. Time in seconds.
+ </para></description>
+ </configOption>
+ <configOption name="max_contacts" default="0">
+ <synopsis>Maximum number of contacts that can bind to an AoR</synopsis>
+ <description><para>
+ Maximum number of contacts that can associate with this AoR. This value does
+ not affect the number of contacts that can be added with the "contact" option.
+ It only limits contacts added through external interaction, such as
+ registration.
+ </para>
+ <note><para>The <replaceable>rewrite_contact</replaceable> option
+ registers the source address as the contact address to help with
+ NAT and reusing connection oriented transports such as TCP and
+ TLS. Unfortunately, refreshing a registration may register a
+ different contact address and exceed
+ <replaceable>max_contacts</replaceable>. The
+ <replaceable>remove_existing</replaceable> and
+ <replaceable>remove_unavailable</replaceable> options can help by
+ removing either the soonest to expire or unavailable contact(s) over
+ <replaceable>max_contacts</replaceable> which is likely the
+ old <replaceable>rewrite_contact</replaceable> contact source
+ address being refreshed.
+ </para></note>
+ <note><para>This should be set to <literal>1</literal> and
+ <replaceable>remove_existing</replaceable> set to <literal>yes</literal> if you
+ wish to stick with the older <literal>chan_sip</literal> behaviour.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="minimum_expiration" default="60">
+ <synopsis>Minimum keep alive time for an AoR</synopsis>
+ <description><para>
+ Minimum time to keep a peer with an explicit expiration. Time in seconds.
+ </para></description>
+ </configOption>
+ <configOption name="remove_existing" default="no">
+ <synopsis>Determines whether new contacts replace existing ones.</synopsis>
+ <description><para>
+ On receiving a new registration to the AoR should it remove enough
+ existing contacts not added or updated by the registration to
+ satisfy <replaceable>max_contacts</replaceable>? Any removed
+ contacts will expire the soonest.
+ </para>
+ <note><para>The <replaceable>rewrite_contact</replaceable> option
+ registers the source address as the contact address to help with
+ NAT and reusing connection oriented transports such as TCP and
+ TLS. Unfortunately, refreshing a registration may register a
+ different contact address and exceed
+ <replaceable>max_contacts</replaceable>. The
+ <replaceable>remove_existing</replaceable> option can help by
+ removing the soonest to expire contact(s) over
+ <replaceable>max_contacts</replaceable> which is likely the
+ old <replaceable>rewrite_contact</replaceable> contact source
+ address being refreshed.
+ </para></note>
+ <note><para>This should be set to <literal>yes</literal> and
+ <replaceable>max_contacts</replaceable> set to <literal>1</literal> if you
+ wish to stick with the older <literal>chan_sip</literal> behaviour.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="remove_unavailable" default="no">
+ <synopsis>Determines whether new contacts should replace unavailable ones.</synopsis>
+ <description><para>
+ The effect of this setting depends on the setting of
+ <replaceable>remove_existing</replaceable>.</para>
+ <para>If <replaceable>remove_existing</replaceable> is set to
+ <literal>no</literal> (default), setting remove_unavailable to
+ <literal>yes</literal> will remove only unavailable contacts that exceed
+ <replaceable>max_contacts</replaceable> to allow an incoming
+ REGISTER to complete sucessfully.</para>
+ <para>If <replaceable>remove_existing</replaceable> is set to
+ <literal>yes</literal>, setting remove_unavailable to
+ <literal>yes</literal> will prioritize unavailable contacts for removal
+ instead of just removing the contact that expires the soonest.</para>
+ <note><para>See <replaceable>remove_existing</replaceable> and
+ <replaceable>max_contacts</replaceable> for further information about how
+ these 3 settings interact.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be of type 'aor'.</synopsis>
+ </configOption>
+ <configOption name="qualify_frequency" default="0">
+ <synopsis>Interval at which to qualify an AoR</synopsis>
+ <description><para>
+ Interval between attempts to qualify the AoR for reachability.
+ If <literal>0</literal> never qualify. Time in seconds.
+ </para></description>
+ </configOption>
+ <configOption name="qualify_timeout" default="3.0">
+ <synopsis>Timeout for qualify</synopsis>
+ <description><para>
+ If the contact doesn't respond to the OPTIONS request before the timeout,
+ the contact is marked unavailable.
+ If <literal>0</literal> no timeout. Time in fractional seconds.
+ </para></description>
+ </configOption>
+ <configOption name="authenticate_qualify">
+ <synopsis>Authenticates a qualify challenge response if needed</synopsis>
+ <description>
+ <para>If true and a qualify request receives a challenge response then
+ authentication is attempted before declaring the contact available.
+ </para>
+ <note><para>This option does nothing as we will always complete
+ the challenge response authentication if the qualify request is
+ challenged.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="outbound_proxy">
+ <synopsis>Outbound proxy used when sending OPTIONS request</synopsis>
+ <description><para>
+ If set the provided URI will be used as the outbound proxy when an
+ OPTIONS request is sent to a contact for qualify purposes.
+ </para></description>
+ </configOption>
+ <configOption name="support_path">
+ <synopsis>Enables Path support for REGISTER requests and Route support for other requests.</synopsis>
+ <description><para>
+ When this option is enabled, the Path headers in register requests will be saved
+ and its contents will be used in Route headers for outbound out-of-dialog requests
+ and in Path headers for outbound 200 responses. Path support will also be indicated
+ in the Supported header.
+ </para></description>
+ </configOption>
+ </configObject>
+ <configObject name="system">
+ <synopsis>Options that apply to the SIP stack as well as other system-wide settings</synopsis>
+ <description><para>
+ The settings in this section are global. In addition to being global, the values will
+ not be re-evaluated when a reload is performed. This is because the values must be set
+ before the SIP stack is initialized. The only way to reset these values is to either
+ restart Asterisk, or unload res_pjsip.so and then load it again.
+ </para></description>
+ <configOption name="timer_t1" default="500">
+ <synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
+ <description><para>
+ Timer T1 is the base for determining how long to wait before retransmitting
+ requests that receive no response when using an unreliable transport (e.g. UDP).
+ For more information on this timer, see RFC 3261, Section 17.1.1.1.
+ </para></description>
+ </configOption>
+ <configOption name="timer_b" default="32000">
+ <synopsis>Set transaction timer B value (milliseconds).</synopsis>
+ <description><para>
+ Timer B determines the maximum amount of time to wait after sending an INVITE
+ request before terminating the transaction. It is recommended that this be set
+ to 64 * Timer T1, but it may be set higher if desired. For more information on
+ this timer, see RFC 3261, Section 17.1.1.1.
+ </para></description>
+ </configOption>
+ <configOption name="compact_headers" default="no">
+ <synopsis>Use the short forms of common SIP header names.</synopsis>
+ </configOption>
+ <configOption name="threadpool_initial_size" default="0">
+ <synopsis>Initial number of threads in the res_pjsip threadpool.</synopsis>
+ </configOption>
+ <configOption name="threadpool_auto_increment" default="5">
+ <synopsis>The amount by which the number of threads is incremented when necessary.</synopsis>
+ </configOption>
+ <configOption name="threadpool_idle_timeout" default="60">
+ <synopsis>Number of seconds before an idle thread should be disposed of.</synopsis>
+ </configOption>
+ <configOption name="threadpool_max_size" default="0">
+ <synopsis>Maximum number of threads in the res_pjsip threadpool.
+ A value of 0 indicates no maximum.</synopsis>
+ </configOption>
+ <configOption name="disable_tcp_switch" default="yes">
+ <synopsis>Disable automatic switching from UDP to TCP transports.</synopsis>
+ <description><para>
+ Disable automatic switching from UDP to TCP transports if outgoing
+ request is too large. See RFC 3261 section 18.1.1.
+ </para></description>
+ </configOption>
+ <configOption name="follow_early_media_fork">
+ <synopsis>Follow SDP forked media when To tag is different</synopsis>
+ <description><para>
+ On outgoing calls, if the UAS responds with different SDP attributes
+ on subsequent 18X or 2XX responses (such as a port update) AND the
+ To tag on the subsequent response is different than that on the previous
+ one, follow it.
+ </para>
+ <note><para>
+ This option must also be enabled on endpoints that require
+ this functionality.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="accept_multiple_sdp_answers">
+ <synopsis>Follow SDP forked media when To tag is the same</synopsis>
+ <description><para>
+ On outgoing calls, if the UAS responds with different SDP attributes
+ on non-100rel 18X or 2XX responses (such as a port update) AND the
+ To tag on the subsequent response is the same as that on the previous one,
+ process the updated SDP.
+ </para>
+ <note><para>
+ This option must also be enabled on endpoints that require
+ this functionality.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="disable_rport" default="no">
+ <synopsis>Disable the use of rport in outgoing requests.</synopsis>
+ <description><para>
+ Remove "rport" parameter from the outgoing requests.
+ </para></description>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be of type 'system' UNLESS the object name is 'system'.</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="global">
+ <synopsis>Options that apply globally to all SIP communications</synopsis>
+ <description><para>
+ The settings in this section are global. Unlike options in the <literal>system</literal>
+ section, these options can be refreshed by performing a reload.
+ </para></description>
+ <configOption name="max_forwards" default="70">
+ <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis>
+ </configOption>
+ <configOption name="keep_alive_interval" default="90">
+ <synopsis>The interval (in seconds) to send keepalives to active connection-oriented transports.</synopsis>
+ </configOption>
+ <configOption name="contact_expiration_check_interval" default="30">
+ <synopsis>The interval (in seconds) to check for expired contacts.</synopsis>
+ </configOption>
+ <configOption name="disable_multi_domain" default="no">
+ <synopsis>Disable Multi Domain support</synopsis>
+ <description><para>
+ If disabled it can improve realtime performance by reducing the number of database requests.
+ </para></description>
+ </configOption>
+ <configOption name="max_initial_qualify_time" default="0">
+ <synopsis>The maximum amount of time from startup that qualifies should be attempted on all contacts.
+ If greater than the qualify_frequency for an aor, qualify_frequency will be used instead.</synopsis>
+ </configOption>
+ <configOption name="unidentified_request_period" default="5">
+ <synopsis>The number of seconds over which to accumulate unidentified requests.</synopsis>
+ <description><para>
+ If <literal>unidentified_request_count</literal> unidentified requests are received
+ during <literal>unidentified_request_period</literal>, a security event will be generated.
+ </para></description>
+ </configOption>
+ <configOption name="unidentified_request_count" default="5">
+ <synopsis>The number of unidentified requests from a single IP to allow.</synopsis>
+ <description><para>
+ If <literal>unidentified_request_count</literal> unidentified requests are received
+ during <literal>unidentified_request_period</literal>, a security event will be generated.
+ </para></description>
+ </configOption>
+ <configOption name="unidentified_request_prune_interval" default="30">
+ <synopsis>The interval at which unidentified requests are older than
+ twice the unidentified_request_period are pruned.</synopsis>
+ </configOption>
+ <configOption name="type">
+ <synopsis>Must be of type 'global' UNLESS the object name is 'global'.</synopsis>
+ </configOption>
+ <configOption name="user_agent" default="Asterisk <Asterisk Version>">
+ <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
+ </configOption>
+ <configOption name="regcontext" default="">
+ <synopsis>When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given
+ peer who registers or unregisters with us.</synopsis>
+ </configOption>
+ <configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
+ <synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
+ </configOption>
+ <configOption name="default_voicemail_extension">
+ <synopsis>The voicemail extension to send in the NOTIFY Message-Account header if not specified on endpoint or aor</synopsis>
+ </configOption>
+ <configOption name="debug" default="no">
+ <synopsis>Enable/Disable SIP debug logging. Valid options include yes, no, or
+ a host address</synopsis>
+ </configOption>
+ <configOption name="endpoint_identifier_order">
+ <synopsis>The order by which endpoint identifiers are processed and checked.
+ Identifier names are usually derived from and can be found in the endpoint
+ identifier module itself (res_pjsip_endpoint_identifier_*).
+ You can use the CLI command "pjsip show identifiers" to see the
+ identifiers currently available.</synopsis>
+ <description>
+ <note><para>
+ One of the identifiers is "auth_username" which matches on the username in
+ an Authentication header. This method has some security considerations because an
+ Authentication header is not present on the first message of a dialog when
+ digest authentication is used. The client can't generate it until the server
+ sends the challenge in a 401 response. Since Asterisk normally sends a security
+ event when an incoming request can't be matched to an endpoint, using auth_username
+ requires that the security event be deferred until a request is received with
+ the Authentication header and only generated if the username doesn't result in a
+ match. This may result in a delay before an attack is recognized. You can control
+ how many unmatched requests are received from a single ip address before a security
+ event is generated using the unidentified_request parameters.
+ </para></note>
+ </description>
+ </configOption>
+ <configOption name="default_from_user" default="asterisk">
+ <synopsis>When Asterisk generates an outgoing SIP request, the From header username will be
+ set to this value if there is no better option (such as CallerID) to be
+ used.</synopsis>
+ </configOption>
+ <configOption name="default_realm" default="asterisk">
+ <synopsis>When Asterisk generates a challenge, the digest realm will be
+ set to this value if there is no better option (such as auth/realm) to be
+ used.</synopsis>
+ </configOption>
+ <configOption name="mwi_tps_queue_high" default="500">
+ <synopsis>MWI taskprocessor high water alert trigger level.</synopsis>
+ <description>
+ <para>On a heavily loaded system you may need to adjust the
+ taskprocessor queue limits. If any taskprocessor queue size
+ reaches its high water level then pjsip will stop processing
+ new requests until the alert is cleared. The alert clears
+ when all alerting taskprocessor queues have dropped to their
+ low water clear level.
+ </para>
+ </description>
+ </configOption>
+ <configOption name="mwi_tps_queue_low" default="-1">
+ <synopsis>MWI taskprocessor low water clear alert level.</synopsis>
+ <description>
+ <para>On a heavily loaded system you may need to adjust the
+ taskprocessor queue limits. If any taskprocessor queue size
+ reaches its high water level then pjsip will stop processing
+ new requests until the alert is cleared. The alert clears
+ when all alerting taskprocessor queues have dropped to their
+ low water clear level.
+ </para>
+ <note><para>Set to -1 for the low water level to be 90% of
+ the high water level.</para></note>
+ </description>
+ </configOption>
+ <configOption name="mwi_disable_initial_unsolicited" default="no">
+ <synopsis>Enable/Disable sending unsolicited MWI to all endpoints on startup.</synopsis>
+ <description>
+ <para>When the initial unsolicited MWI notification are
+ enabled on startup then the initial notifications
+ get sent at startup. If you have a lot of endpoints
+ (thousands) that use unsolicited MWI then you may
+ want to consider disabling the initial startup
+ notifications.
+ </para>
+ <para>When the initial unsolicited MWI notifications are
+ disabled on startup then the notifications will start
+ on the endpoint's next contact update.
+ </para>
+ </description>
+ </configOption>
+ <configOption name="ignore_uri_user_options">
+ <synopsis>Enable/Disable ignoring SIP URI user field options.</synopsis>
+ <description>
+ <para>If you have this option enabled and there are semicolons
+ in the user field of a SIP URI then the field is truncated
+ at the first semicolon. This effectively makes the semicolon
+ a non-usable character for PJSIP endpoint names, extensions,
+ and AORs. This can be useful for improving compatibility with
+ an ITSP that likes to use user options for whatever reason.
+ </para>
+ <example title="Sample SIP URI">
+ sip:1235557890;phone-context=national@x.x.x.x;user=phone
+ </example>
+ <example title="Sample SIP URI user field">
+ 1235557890;phone-context=national
+ </example>
+ <example title="Sample SIP URI user field truncated">
+ 1235557890
+ </example>
+ <note><para>The caller-id and redirecting number strings
+ obtained from incoming SIP URI user fields are always truncated
+ at the first semicolon.</para></note>
+ </description>
+ </configOption>
+ <configOption name="use_callerid_contact" default="no">
+ <synopsis>Place caller-id information into Contact header</synopsis>
+ <description><para>
+ This option will cause Asterisk to place caller-id information into
+ generated Contact headers.</para>
+ </description>
+ </configOption>
+ <configOption name="send_contact_status_on_update_registration" default="no">
+ <synopsis>Enable sending AMI ContactStatus event when a device refreshes its registration.</synopsis>
+ </configOption>
+ <configOption name="taskprocessor_overload_trigger">
+ <synopsis>Trigger scope for taskprocessor overloads</synopsis>
+ <description><para>
+ This option specifies the trigger the distributor will use for
+ detecting taskprocessor overloads. When it detects an overload condition,
+ the distrubutor will stop accepting new requests until the overload is
+ cleared.
+ </para>
+ <enumlist>
+ <enum name="global"><para>(default) Any taskprocessor overload will trigger.</para></enum>
+ <enum name="pjsip_only"><para>Only pjsip taskprocessor overloads will trigger.</para></enum>
+ <enum name="none"><para>No overload detection will be performed.</para></enum>
+ </enumlist>
+ <warning><para>
+ The "none" and "pjsip_only" options should be used
+ with extreme caution and only to mitigate specific issues.
+ Under certain conditions they could make things worse.
+ </para></warning>
+ </description>
+ </configOption>
+ <configOption name="norefersub" default="yes">
+ <synopsis>Advertise support for RFC4488 REFER subscription suppression</synopsis>
+ </configOption>
+ </configObject>
+ </configFile>
+ </configInfo>
+</docs>
--- /dev/null
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE docs SYSTEM "appdocsxml.dtd">
+<docs>
+ <manager name="PJSIPQualify" language="en_US">
+ <synopsis>
+ Qualify a chan_pjsip endpoint.
+ </synopsis>
+ <syntax>
+ <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
+ <parameter name="Endpoint" required="true">
+ <para>The endpoint you want to qualify.</para>
+ </parameter>
+ </syntax>
+ <description>
+ <para>Qualify a chan_pjsip endpoint.</para>
+ </description>
+ </manager>
+ <managerEvent language="en_US" name="IdentifyDetail">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide details about an identify section.</synopsis>
+ <syntax>
+ <parameter name="ObjectType">
+ <para>The object's type. This will always be 'identify'.</para>
+ </parameter>
+ <parameter name="ObjectName">
+ <para>The name of this object.</para>
+ </parameter>
+ <parameter name="Endpoint">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='endpoint']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SrvLookups">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='srv_lookups']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Match">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='match']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MatchHeader">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip_endpoint_identifier_ip']/configFile[@name='pjsip.conf']/configObject[@name='identify']/configOption[@name='match_header']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="EndpointName">
+ <para>The name of the endpoint associated with this information.</para>
+ </parameter>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ <managerEvent language="en_US" name="AorDetail">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide details about an Address of Record (AoR) section.</synopsis>
+ <syntax>
+ <parameter name="ObjectType">
+ <para>The object's type. This will always be 'aor'.</para>
+ </parameter>
+ <parameter name="ObjectName">
+ <para>The name of this object.</para>
+ </parameter>
+ <parameter name="MinimumExpiration">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='minimum_expiration']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MaximumExpiration">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='maximum_expiration']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DefaultExpiration">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='default_expiration']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="QualifyFrequency">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='qualify_frequency']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="AuthenticateQualify">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='authenticate_qualify']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MaxContacts">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='max_contacts']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RemoveExisting">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='remove_existing']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RemoveUnavailable">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='remove_unavailable']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Mailboxes">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='mailboxes']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="OutboundProxy">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='outbound_proxy']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SupportPath">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='support_path']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="TotalContacts">
+ <para>The total number of contacts associated with this AoR.</para>
+ </parameter>
+ <parameter name="ContactsRegistered">
+ <para>The number of non-permanent contacts associated with this AoR.</para>
+ </parameter>
+ <parameter name="EndpointName">
+ <para>The name of the endpoint associated with this information.</para>
+ </parameter>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ <managerEvent language="en_US" name="AuthDetail">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide details about an authentication section.</synopsis>
+ <syntax>
+ <parameter name="ObjectType">
+ <para>The object's type. This will always be 'auth'.</para>
+ </parameter>
+ <parameter name="ObjectName">
+ <para>The name of this object.</para>
+ </parameter>
+ <parameter name="Username">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='username']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Password">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='username']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Md5Cred">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='md5_cred']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Realm">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='realm']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="NonceLifetime">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='nonce_lifetime']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="AuthType">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='auth_type']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="EndpointName">
+ <para>The name of the endpoint associated with this information.</para>
+ </parameter>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ <managerEvent language="en_US" name="TransportDetail">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide details about an authentication section.</synopsis>
+ <syntax>
+ <parameter name="ObjectType">
+ <para>The object's type. This will always be 'transport'.</para>
+ </parameter>
+ <parameter name="ObjectName">
+ <para>The name of this object.</para>
+ </parameter>
+ <parameter name="Protocol">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='protocol']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Bind">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='bind']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="AsycOperations">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='async_operations']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="CaListFile">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='ca_list_file']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="CaListPath">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='ca_list_path']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="CertFile">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cert_file']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="PrivKeyFile">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='priv_key_file']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Password">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='password']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="ExternalSignalingAddress">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_signaling_address']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="ExternalSignalingPort">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_signaling_port']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="ExternalMediaAddress">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='external_media_address']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Domain">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='domain']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="VerifyServer">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='verify_server']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="VerifyClient">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='verify_client']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RequireClientCert">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='require_client_cert']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Method">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='method']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Cipher">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cipher']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="LocalNet">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='local_net']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Tos">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='tos']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Cos">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='cos']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="WebsocketWriteTimeout">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='transport']/configOption[@name='websocket_write_timeout']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="EndpointName">
+ <para>The name of the endpoint associated with this information.</para>
+ </parameter>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ <managerEvent language="en_US" name="EndpointDetail">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide details about an endpoint section.</synopsis>
+ <syntax>
+ <parameter name="ObjectType">
+ <para>The object's type. This will always be 'endpoint'.</para>
+ </parameter>
+ <parameter name="ObjectName">
+ <para>The name of this object.</para>
+ </parameter>
+ <parameter name="Context">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='context']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Disallow">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='disallow']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Allow">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DtmfMode">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtmf_mode']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RtpIpv6">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_ipv6']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RtpSymmetric">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_symmetric']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="IceSupport">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='ice_support']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="UsePtime">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='use_ptime']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="ForceRport">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='force_rport']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RewriteContact">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rewrite_contact']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Transport">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='transport']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="OutboundProxy">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='outbound_proxy']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MohSuggest">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='moh_suggest']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="100rel">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='100rel']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Timers">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="TimersMinSe">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers_min_se']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="TimersSessExpires">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='timers_sess_expires']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Auth">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='auth']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="OutboundAuth">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='outbound_auth']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Aors">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='aors']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MediaAddress">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_address']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="IdentifyBy">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='identify_by']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DirectMedia">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DirectMediaMethod">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media_method']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="TrustConnectedLine">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_connected_line']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SendConnectedLine">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_connected_line']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="ConnectedLineMethod">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='connected_line_method']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DirectMediaGlareMitigation">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='direct_media_glare_mitigation']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DisableDirectMediaOnNat">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='disable_direct_media_on_nat']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Callerid">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="CalleridPrivacy">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid_privacy']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="CalleridTag">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='callerid_tag']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="TrustIdInbound">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_id_inbound']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="TrustIdOutbound">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='trust_id_outbound']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SendPai">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_pai']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SendRpid">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_rpid']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SendDiversion">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='send_diversion']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Mailboxes">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='mailboxes']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="AggregateMwi">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='aggregate_mwi']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MediaEncryption">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_encryption']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MediaEncryptionOptimistic">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_encryption_optimistic']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="UseAvpf">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='use_avpf']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="ForceAvp">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='force_avp']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MediaUseReceivedTransport">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='media_use_received_transport']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="OneTouchRecording">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='one_touch_recording']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="InbandProgress">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='inband_progress']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="CallGroup">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='call_group']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="PickupGroup">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='pickup_group']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="NamedCallGroup">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='named_call_group']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="NamedPickupGroup">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='named_pickup_group']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DeviceStateBusyAt">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='device_state_busy_at']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="T38Udptl">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="T38UdptlEc">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_ec']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="T38UdptlMaxdatagram">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_maxdatagram']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="FaxDetect">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='fax_detect']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="T38UdptlNat">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_nat']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="T38UdptlIpv6">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_udptl_ipv6']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="T38BindUdptlToMediaAddress">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='t38_bind_udptl_to_media_address']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="ToneZone">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tone_zone']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Language">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='language']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RecordOnFeature">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='record_on_feature']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RecordOffFeature">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='record_off_feature']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="AllowTransfer">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_transfer']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="UserEqPhone">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='user_eq_phone']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MohPassthrough">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='moh_passthrough']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SdpOwner">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_owner']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SdpSession">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_session']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="TosAudio">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tos_audio']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="TosVideo">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='tos_video']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="CosAudio">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='cos_audio']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="CosVideo">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='cos_video']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="AllowSubscribe">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_subscribe']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SubMinExpiry">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sub_min_expiry']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="FromUser">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='from_user']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="FromDomain">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='from_domain']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MwiFromUser">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='mwi_from_user']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RtpEngine">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rtp_engine']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DtlsVerify">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_verify']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DtlsRekey">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_rekey']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DtlsCertFile">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_cert_file']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DtlsPrivateKey">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_private_key']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DtlsCipher">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_cipher']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DtlsCaFile">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_ca_file']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DtlsCaPath">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_ca_path']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DtlsSetup">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='dtls_setup']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SrtpTag32">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='srtp_tag_32']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RedirectMethod">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='redirect_method']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SetVar">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='set_var']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MessageContext">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='message_context']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Accountcode">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='accountcode']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="PreferredCodecOnly">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='preferred_codec_only']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DeviceState">
+ <para>The aggregate device state for this endpoint.</para>
+ </parameter>
+ <parameter name="ActiveChannels">
+ <para>The number of active channels associated with this endpoint.</para>
+ </parameter>
+ <parameter name="SubscribeContext">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='subscribe_context']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Allowoverlap">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='allow_overlap']/synopsis/node())"/></para>
+ </parameter>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ <managerEvent language="en_US" name="AorList">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide details about an Address of Record (AoR) section.</synopsis>
+ <syntax>
+ <parameter name="ObjectType">
+ <para>The object's type. This will always be 'aor'.</para>
+ </parameter>
+ <parameter name="ObjectName">
+ <para>The name of this object.</para>
+ </parameter>
+ <parameter name="MinimumExpiration">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='minimum_expiration']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MaximumExpiration">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='maximum_expiration']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="DefaultExpiration">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='default_expiration']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="QualifyFrequency">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='qualify_frequency']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="AuthenticateQualify">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='authenticate_qualify']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="MaxContacts">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='max_contacts']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RemoveExisting">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='remove_existing']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="RemoveUnavailable">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='remove_unavailable']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Mailboxes">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='mailboxes']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="OutboundProxy">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='outbound_proxy']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="SupportPath">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='aor']/configOption[@name='support_path']/synopsis/node())"/></para>
+ </parameter>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ <managerEvent language="en_US" name="AuthList">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide details about an Address of Record (Auth) section.</synopsis>
+ <syntax>
+ <parameter name="ObjectType">
+ <para>The object's type. This will always be 'auth'.</para>
+ </parameter>
+ <parameter name="ObjectName">
+ <para>The name of this object.</para>
+ </parameter>
+ <parameter name="Username">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='username']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Md5Cred">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='md5_cred']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Realm">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='realm']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="AuthType">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='auth_type']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="Password">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='password']/synopsis/node())"/></para>
+ </parameter>
+ <parameter name="NonceLifetime">
+ <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='auth']/configOption[@name='nonce_lifetime']/synopsis/node())"/></para>
+ </parameter>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ <managerEvent language="en_US" name="ContactList">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide details about a contact section.</synopsis>
+ <syntax>
+ <parameter name="ObjectType">
+ <para>The object's type. This will always be 'contact'.</para>
+ </parameter>
+ <parameter name="ObjectName">
+ <para>The name of this object.</para>
+ </parameter>
+ <parameter name="ViaAddr">
+ <para>IP address of the last Via header in REGISTER request.
+ Will only appear in the event if available.</para>
+ </parameter>
+ <parameter name="ViaPort">
+ <para>Port number of the last Via header in REGISTER request.
+ Will only appear in the event if available.</para>
+ </parameter>
+ <parameter name="QualifyTimeout">
+ <para>The elapsed time in decimal seconds after which an OPTIONS
+ message is sent before the contact is considered unavailable.</para>
+ </parameter>
+ <parameter name="CallId">
+ <para>Content of the Call-ID header in REGISTER request.
+ Will only appear in the event if available.</para>
+ </parameter>
+ <parameter name="RegServer">
+ <para>Asterisk Server name.</para>
+ </parameter>
+ <parameter name="PruneOnBoot">
+ <para>If true delete the contact on Asterisk restart/boot.</para>
+ </parameter>
+ <parameter name="Path">
+ <para>The Path header received on the REGISTER.</para>
+ </parameter>
+ <parameter name="Endpoint">
+ <para>The name of the endpoint associated with this information.</para>
+ </parameter>
+ <parameter name="AuthenticateQualify">
+ <para>A boolean indicating whether a qualify should be authenticated.</para>
+ </parameter>
+ <parameter name="Uri">
+ <para>This contact's URI.</para>
+ </parameter>
+ <parameter name="QualifyFrequency">
+ <para>The interval in seconds at which the contact will be qualified.</para>
+ </parameter>
+ <parameter name="UserAgent">
+ <para>Content of the User-Agent header in REGISTER request</para>
+ </parameter>
+ <parameter name="ExpirationTime">
+ <para>Absolute time that this contact is no longer valid after</para>
+ </parameter>
+ <parameter name="OutboundProxy">
+ <para>The contact's outbound proxy.</para>
+ </parameter>
+ <parameter name="Status">
+ <para>This contact's status.</para>
+ <enumlist>
+ <enum name="Reachable"/>
+ <enum name="Unreachable"/>
+ <enum name="NonQualified"/>
+ <enum name="Unknown"/>
+ </enumlist>
+ </parameter>
+ <parameter name="RoundtripUsec">
+ <para>The round trip time in microseconds.</para>
+ </parameter>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ <managerEvent language="en_US" name="ContactStatusDetail">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide details about a contact's status.</synopsis>
+ <syntax>
+ <parameter name="AOR">
+ <para>The AoR that owns this contact.</para>
+ </parameter>
+ <parameter name="URI">
+ <para>This contact's URI.</para>
+ </parameter>
+ <parameter name="Status">
+ <para>This contact's status.</para>
+ <enumlist>
+ <enum name="Reachable"/>
+ <enum name="Unreachable"/>
+ <enum name="NonQualified"/>
+ <enum name="Unknown"/>
+ </enumlist>
+ </parameter>
+ <parameter name="RoundtripUsec">
+ <para>The round trip time in microseconds.</para>
+ </parameter>
+ <parameter name="EndpointName">
+ <para>The name of the endpoint associated with this information.</para>
+ </parameter>
+ <parameter name="UserAgent">
+ <para>Content of the User-Agent header in REGISTER request</para>
+ </parameter>
+ <parameter name="RegExpire">
+ <para>Absolute time that this contact is no longer valid after</para>
+ </parameter>
+ <parameter name="ViaAddress">
+ <para>IP address:port of the last Via header in REGISTER request.
+ Will only appear in the event if available.</para>
+ </parameter>
+ <parameter name="CallID">
+ <para>Content of the Call-ID header in REGISTER request.
+ Will only appear in the event if available.</para>
+ </parameter>
+ <parameter name="ID">
+ <para>The sorcery ID of the contact.</para>
+ </parameter>
+ <parameter name="AuthenticateQualify">
+ <para>A boolean indicating whether a qualify should be authenticated.</para>
+ </parameter>
+ <parameter name="OutboundProxy">
+ <para>The contact's outbound proxy.</para>
+ </parameter>
+ <parameter name="Path">
+ <para>The Path header received on the REGISTER.</para>
+ </parameter>
+ <parameter name="QualifyFrequency">
+ <para>The interval in seconds at which the contact will be qualified.</para>
+ </parameter>
+ <parameter name="QualifyTimeout">
+ <para>The elapsed time in decimal seconds after which an OPTIONS
+ message is sent before the contact is considered unavailable.</para>
+ </parameter>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ <managerEvent language="en_US" name="EndpointList">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide details about a contact's status.</synopsis>
+ <syntax>
+ <parameter name="ObjectType">
+ <para>The object's type. This will always be 'endpoint'.</para>
+ </parameter>
+ <parameter name="ObjectName">
+ <para>The name of this object.</para>
+ </parameter>
+ <parameter name="Transport">
+ <para>The transport configurations associated with this endpoint.</para>
+ </parameter>
+ <parameter name="Aor">
+ <para>The aor configurations associated with this endpoint.</para>
+ </parameter>
+ <parameter name="Auths">
+ <para>The inbound authentication configurations associated with this endpoint.</para>
+ </parameter>
+ <parameter name="OutboundAuths">
+ <para>The outbound authentication configurations associated with this endpoint.</para>
+ </parameter>
+ <parameter name="DeviceState">
+ <para>The aggregate device state for this endpoint.</para>
+ </parameter>
+ <parameter name="ActiveChannels">
+ <para>The number of active channels associated with this endpoint.</para>
+ </parameter>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ <manager name="PJSIPShowEndpoints" language="en_US">
+ <synopsis>
+ Lists PJSIP endpoints.
+ </synopsis>
+ <syntax />
+ <description>
+ <para>
+ Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
+ is raised that contains relevant attributes and status information. Once all
+ endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
+ </para>
+ </description>
+ <responses>
+ <list-elements>
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='EndpointList'])" />
+ </list-elements>
+ <managerEvent language="en_US" name="EndpointListComplete">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide final information about an endpoint list.</synopsis>
+ <syntax>
+ <parameter name="EventList"/>
+ <parameter name="ListItems"/>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ </responses>
+ </manager>
+ <manager name="PJSIPShowEndpoint" language="en_US">
+ <synopsis>
+ Detail listing of an endpoint and its objects.
+ </synopsis>
+ <syntax>
+ <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
+ <parameter name="Endpoint" required="true">
+ <para>The endpoint to list.</para>
+ </parameter>
+ </syntax>
+ <description>
+ <para>
+ Provides a detailed listing of options for a given endpoint. Events are issued
+ showing the configuration and status of the endpoint and associated objects. These
+ events include <literal>EndpointDetail</literal>, <literal>AorDetail</literal>,
+ <literal>AuthDetail</literal>, <literal>TransportDetail</literal>, and
+ <literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
+ associated (for instance AoRs). Once all detail events have been raised a final
+ <literal>EndpointDetailComplete</literal> event is issued.
+ </para>
+ </description>
+ <responses>
+ <list-elements>
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='EndpointDetail'])" />
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='IdentifyDetail'])" />
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='ContactStatusDetail'])" />
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='AuthDetail'])" />
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='TransportDetail'])" />
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='AorDetail'])" />
+ </list-elements>
+ <managerEvent language="en_US" name="EndpointDetailComplete">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide final information about endpoint details.</synopsis>
+ <syntax>
+ <parameter name="EventList"/>
+ <parameter name="ListItems"/>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ </responses>
+ </manager>
+ <manager name="PJSIPShowAors" language="en_US">
+ <synopsis>
+ Lists PJSIP AORs.
+ </synopsis>
+ <syntax />
+ <description>
+ <para>
+ Provides a listing of all AORs. For each AOR an <literal>AorList</literal> event
+ is raised that contains relevant attributes and status information. Once all
+ aors have been listed an <literal>AorListComplete</literal> event is issued.
+ </para>
+ </description>
+ <responses>
+ <list-elements>
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='AorList'])" />
+ </list-elements>
+ <managerEvent language="en_US" name="AorListComplete">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide final information about an aor list.</synopsis>
+ <syntax>
+ <parameter name="EventList"/>
+ <parameter name="ListItems"/>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ </responses>
+ </manager>
+ <manager name="PJSIPShowAuths" language="en_US">
+ <synopsis>
+ Lists PJSIP Auths.
+ </synopsis>
+ <syntax />
+ <description>
+ <para>Provides a listing of all Auths. For each Auth an <literal>AuthList</literal> event
+ is raised that contains relevant attributes and status information. Once all
+ auths have been listed an <literal>AuthListComplete</literal> event is issued.
+ </para>
+ </description>
+ <responses>
+ <list-elements>
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='AuthList'])" />
+ </list-elements>
+ <managerEvent language="en_US" name="AuthListComplete">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide final information about an auth list.</synopsis>
+ <syntax>
+ <parameter name="EventList"/>
+ <parameter name="ListItems"/>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ </responses>
+ </manager>
+ <manager name="PJSIPShowContacts" language="en_US">
+ <synopsis>
+ Lists PJSIP Contacts.
+ </synopsis>
+ <syntax />
+ <description>
+ <para>Provides a listing of all Contacts. For each Contact a <literal>ContactList</literal>
+ event is raised that contains relevant attributes and status information.
+ Once all contacts have been listed a <literal>ContactListComplete</literal> event
+ is issued.
+ </para>
+ </description>
+ <responses>
+ <list-elements>
+ <xi:include xpointer="xpointer(/docs/managerEvent[@name='ContactList'])" />
+ </list-elements>
+ <managerEvent language="en_US" name="ContactListComplete">
+ <managerEventInstance class="EVENT_FLAG_COMMAND">
+ <synopsis>Provide final information about a contact list.</synopsis>
+ <syntax>
+ <parameter name="EventList"/>
+ <parameter name="ListItems"/>
+ </syntax>
+ </managerEventInstance>
+ </managerEvent>
+ </responses>
+ </manager>
+</docs>
projects / thirdparty / asterisk.git / commitdiff
