+++ /dev/null
-<profile name="external">
- <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
- <!-- This profile is only for outbound registrations to providers -->
- <gateways>
- <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
- </gateways>
-
- <aliases>
- <!--
- <alias name="outbound"/>
- <alias name="nat"/>
- -->
- </aliases>
-
- <domains>
- <domain name="all" alias="false" parse="true"/>
- </domains>
-
- <settings>
- <param name="debug" value="0"/>
- <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
- <!-- <param name="shutdown-on-fail" value="true"/> -->
- <param name="sip-trace" value="no"/>
- <param name="sip-capture" value="no"/>
- <param name="rfc2833-pt" value="101"/>
- <!-- RFC 5626 : Send reg-id and sip.instance -->
- <!--<param name="enable-rfc-5626" value="true"/> -->
- <param name="sip-port" value="$${external_sip_port}"/>
- <param name="dialplan" value="XML"/>
- <param name="context" value="public"/>
- <param name="dtmf-duration" value="2000"/>
- <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
- <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
- <param name="hold-music" value="$${hold_music}"/>
- <param name="rtp-timer-name" value="soft"/>
- <!--<param name="enable-100rel" value="true"/>-->
- <!--<param name="disable-srv503" value="true"/>-->
- <!-- This could be set to "passive" -->
- <param name="local-network-acl" value="localnet.auto"/>
- <param name="manage-presence" value="false"/>
-
- <!-- used to share presence info across sofia profiles
- manage-presence needs to be set to passive on this profile
- if you want it to behave as if it were the internal profile
- for presence.
- -->
- <!-- Name of the db to use for this profile -->
- <!--<param name="dbname" value="share_presence"/>-->
- <!--<param name="presence-hosts" value="$${domain}"/>-->
- <!--<param name="force-register-domain" value="$${domain}"/>-->
- <!--all inbound reg will stored in the db using this domain -->
- <!--<param name="force-register-db-domain" value="$${domain}"/>-->
- <!-- ************************************************* -->
-
- <!--<param name="aggressive-nat-detection" value="true"/>-->
- <param name="inbound-codec-negotiation" value="generous"/>
- <param name="nonce-ttl" value="60"/>
- <param name="auth-calls" value="false"/>
- <!--
- DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
- -->
- <param name="rtp-ip" value="$${local_ip_v4}"/>
- <param name="sip-ip" value="$${local_ip_v4}"/>
- <param name="ext-rtp-ip" value="auto-nat"/>
- <param name="ext-sip-ip" value="auto-nat"/>
- <param name="rtp-timeout-sec" value="300"/>
- <param name="rtp-hold-timeout-sec" value="1800"/>
- <!--<param name="enable-3pcc" value="true"/>-->
-
- <!-- TLS: disabled by default, set to "true" to enable -->
- <param name="tls" value="$${external_ssl_enable}"/>
- <!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
- <param name="tls-only" value="false"/>
- <!-- additional bind parameters for TLS -->
- <param name="tls-bind-params" value="transport=tls"/>
- <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
- <param name="tls-sip-port" value="$${external_tls_port}"/>
- <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
- <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
- <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
- <param name="tls-passphrase" value=""/>
- <!-- Verify the date on TLS certificates -->
- <param name="tls-verify-date" value="true"/>
- <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
- <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
- <param name="tls-verify-policy" value="none"/>
- <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
- <param name="tls-verify-depth" value="2"/>
- <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
- <param name="tls-verify-in-subjects" value=""/>
- <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
- <param name="tls-version" value="$${sip_tls_version}"/>
-
- </settings>
-</profile>
+++ /dev/null
-<include>
- <!--<gateway name="asterlink.com">-->
- <!--/// account username *required* ///-->
- <!--<param name="username" value="cluecon"/>-->
- <!--/// auth realm: *optional* same as gateway name, if blank ///-->
- <!--<param name="realm" value="asterlink.com"/>-->
- <!--/// username to use in from: *optional* same as username, if blank ///-->
- <!--<param name="from-user" value="cluecon"/>-->
- <!--/// domain to use in from: *optional* same as realm, if blank ///-->
- <!--<param name="from-domain" value="asterlink.com"/>-->
- <!--/// account password *required* ///-->
- <!--<param name="password" value="2007"/>-->
- <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
- <!--<param name="extension" value="cluecon"/>-->
- <!--/// proxy host: *optional* same as realm, if blank ///-->
- <!--<param name="proxy" value="asterlink.com"/>-->
- <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
- <!--<param name="register-proxy" value="mysbc.com"/>-->
- <!--/// expire in seconds: *optional* 3600, if blank ///-->
- <!--<param name="expire-seconds" value="60"/>-->
- <!--/// do not register ///-->
- <!--<param name="register" value="false"/>-->
- <!-- which transport to use for register -->
- <!--<param name="register-transport" value="udp"/>-->
- <!--How many seconds before a retry when a failure or timeout occurs -->
- <!--<param name="retry-seconds" value="30"/>-->
- <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
- <!--<param name="caller-id-in-from" value="false"/>-->
- <!--extra sip params to send in the contact-->
- <!--<param name="contact-params" value="tport=tcp"/>-->
- <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
- <!--<param name="ping" value="25"/>-->
- <!--</gateway>-->
- <!--rfc5626 : Abilitazione rfc5626 ///-->
- <!--<param name="rfc-5626" value="true"/>-->
- <!--rfc5626 : extra sip params to send in the contact-->
- <!--<param name="reg-id" value="1"/>-->
-</include>
+++ /dev/null
-<profile name="internal-ipv6">
- <!--
- This is an example of a sofia profile setup to listen on IPv6.
- -->
- <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
- <!--aliases are other names that will work as a valid profile name for this profile-->
- <settings>
- <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
- <param name="debug" value="0"/>
- <param name="sip-trace" value="no"/>
- <param name="context" value="public"/>
- <param name="rfc2833-pt" value="101"/>
- <!-- port to bind to for sip traffic -->
- <param name="sip-port" value="$${internal_sip_port}"/>
- <param name="dialplan" value="XML"/>
- <param name="dtmf-duration" value="2000"/>
- <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
- <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
- <param name="use-rtp-timer" value="true"/>
- <param name="rtp-timer-name" value="soft"/>
- <!-- ip address to use for rtp -->
- <param name="rtp-ip" value="$${local_ip_v6}"/>
- <!-- ip address to bind to -->
- <param name="sip-ip" value="$${local_ip_v6}"/>
- <param name="hold-music" value="$${hold_music}"/>
- <!--<param name="enable-100rel" value="false"/>-->
- <!--<param name="disable-srv503" value="true"/>-->
- <param name="apply-inbound-acl" value="domains"/>
- <!--<param name="apply-register-acl" value="domains"/>-->
- <!--<param name="dtmf-type" value="info"/>-->
- <param name="record-template" value="$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
- <!--enable to use presence and mwi -->
- <param name="manage-presence" value="true"/>
- <!-- This setting is for AAL2 bitpacking on G726 -->
- <!-- <param name="bitpacking" value="aal2"/> -->
- <!--max number of open dialogs in proceeding -->
- <!--<param name="max-proceeding" value="1000"/>-->
- <!--session timers for all call to expire after the specified seconds -->
- <!--<param name="session-timeout" value="1800"/>-->
- <!--<param name="multiple-registrations" value="true"/>-->
- <!--set to 'greedy' if you want your codec list to take precedence -->
- <param name="inbound-codec-negotiation" value="generous"/>
- <!-- if you want to send any special bind params of your own -->
- <!--<param name="bind-params" value="transport=udp"/>-->
- <!--<param name="unregister-on-options-fail" value="true"/>-->
-
- <!-- TLS: disabled by default, set to "true" to enable -->
- <param name="tls" value="$${internal_ssl_enable}"/>
- <!-- additional bind parameters for TLS -->
- <param name="tls-bind-params" value="transport=tls"/>
- <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
- <param name="tls-sip-port" value="$${internal_tls_port}"/>
- <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
- <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
- <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
- <param name="tls-version" value="$${sip_tls_version}"/>
-
- <!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
- <!--<param name="rtp-rewrite-timestamps" value="true"/>-->
- <!--<param name="pass-rfc2833" value="true"/>-->
- <!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
- <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
-
- <!--Uncomment to set all inbound calls to no media mode-->
- <!--<param name="inbound-bypass-media" value="true"/>-->
-
- <!--Uncomment to set all inbound calls to proxy media mode-->
- <!--<param name="inbound-proxy-media" value="true"/>-->
-
- <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
- <!--<param name="inbound-late-negotiation" value="true"/>-->
-
- <!-- this lets anything register -->
- <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
- <!-- <param name="accept-blind-reg" value="true"/> -->
-
- <!-- accept any authentication without actually checking (not a good feature for most people) -->
- <!-- <param name="accept-blind-auth" value="true"/> -->
-
- <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
- <!-- <param name="suppress-cng" value="true"/> -->
-
- <!--TTL for nonce in sip auth-->
- <param name="nonce-ttl" value="60"/>
- <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
- that the originator is using-->
- <!--<param name="disable-transcoding" value="true"/>-->
- <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
- <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
- <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
- <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
- <param name="auth-calls" value="$${internal_auth_calls}"/>
- <!-- on authed calls, authenticate *all* the packets not just invite -->
- <param name="auth-all-packets" value="false"/>
- <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
- <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
- <!-- rtp inactivity timeout -->
- <param name="rtp-timeout-sec" value="300"/>
- <param name="rtp-hold-timeout-sec" value="1800"/>
- <!-- VAD choose one (out is a good choice); -->
- <!-- <param name="vad" value="in"/> -->
- <!-- <param name="vad" value="out"/> -->
- <!-- <param name="vad" value="both"/> -->
- <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
- <!--
- These are enabled to make the default config work better out of the box.
- If you need more than ONE domain you'll need to not use these options.
-
- -->
- <!--all inbound reg will look in this domain for the users -->
- <param name="force-register-domain" value="$${domain}"/>
- <!--all inbound reg will stored in the db using this domain -->
- <param name="force-register-db-domain" value="$${domain}"/>
- <!-- disable register and transfer which may be undesirable in a public switch -->
- <!--<param name="disable-transfer" value="true"/>-->
- <!--<param name="disable-register" value="true"/>-->
- <!--<param name="enable-3pcc" value="true"/>-->
- <!-- use stun when specified (default is true) -->
- <!--<param name="stun-enabled" value="true"/>-->
- <!-- use stun when specified (default is true) -->
- <!-- set to true to have the profile determine stun is not useful and turn it off globally-->
- <!--<param name="stun-auto-disable" value="true"/>-->
-
- <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
- <!--<param name="disable-srv" value="false" />-->
- <!--<param name="disable-naptr" value="false" />-->
-
- </settings>
-</profile>
-
+++ /dev/null
-<profile name="internal">
- <!--
- This is a sofia sip profile/user agent. This will service exactly one ip and port.
- In FreeSWITCH you can run multiple sip user agents on their own ip and port.
-
- When you hear someone say "sofia profile" this is what they are talking about.
- -->
-
- <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
- <!--aliases are other names that will work as a valid profile name for this profile-->
- <aliases>
- <!--
- <alias name="default"/>
- -->
- </aliases>
- <!-- Outbound Registrations -->
- <gateways>
- <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
- </gateways>
-
- <domains>
- <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
- <!--<domain name="$${domain}" parse="true"/>-->
- <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
- <!--<domain name="all" alias="true" parse="true"/>-->
- <domain name="all" alias="true" parse="false"/>
- </domains>
-
- <settings>
- <!--
- When calls are in no media this will bring them back to media
- when you press the hold button.
- -->
- <!--<param name="media-option" value="resume-media-on-hold"/> -->
- <!--
- This will allow a call after an attended transfer go back to
- bypass media after an attended transfer.
- -->
- <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
- <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
- <param name="debug" value="0"/>
- <!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
- <!-- <param name="shutdown-on-fail" value="true"/> -->
- <param name="sip-trace" value="no"/>
- <param name="sip-capture" value="no"/>
-
- <!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
- <!-- <param name="presence-proto-lookup" value="true"/> -->
-
-
- <!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
- <!--<param name="liberal-dtmf" value="true"/>-->
-
-
- <!--
- Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
- responding. These options allow you to enable and control a watchdog
- on the Sofia SIP stack so that if it stops responding for the
- specified number of milliseconds, it will cause FreeSWITCH to crash
- immediately. This is useful if you run in an HA environment and
- need to ensure automated recovery from such a condition. Note that if
- your server is idle a lot, the watchdog may fire due to not receiving
- any SIP messages. Thus, if you expect your system to be idle, you
- should leave the watchdog disabled. It can be toggled on and off
- through the FreeSWITCH CLI either on an individual profile basis or
- globally for all profiles. So, if you run in an HA environment with a
- master and slave, you should use the CLI to make sure the watchdog is
- only enabled on the master.
- If such crash occurs, FreeSWITCH will dump core if allowed. The
- stacktrace will include function watchdog_triggered_abort().
- -->
- <param name="watchdog-enabled" value="no"/>
- <param name="watchdog-step-timeout" value="30000"/>
- <param name="watchdog-event-timeout" value="30000"/>
-
- <param name="log-auth-failures" value="false"/>
- <param name="forward-unsolicited-mwi-notify" value="false"/>
-
- <param name="context" value="public"/>
- <param name="rfc2833-pt" value="101"/>
- <!-- port to bind to for sip traffic -->
- <param name="sip-port" value="$${internal_sip_port}"/>
- <param name="dialplan" value="XML"/>
- <param name="dtmf-duration" value="2000"/>
- <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
- <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
- <param name="rtp-timer-name" value="soft"/>
- <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
- <param name="rtp-ip" value="$${local_ip_v4}"/>
- <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
- <param name="sip-ip" value="$${local_ip_v4}"/>
- <param name="hold-music" value="$${hold_music}"/>
- <param name="apply-nat-acl" value="nat.auto"/>
-
-
- <!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
- <!-- <param name="cid-in-1xx" value="false"/> -->
-
- <!-- extended info parsing -->
- <!-- <param name="extended-info-parsing" value="true"/> -->
-
- <!--<param name="aggressive-nat-detection" value="true"/>-->
- <!--
- There are known issues (asserts and segfaults) when 100rel is enabled.
- It is not recommended to enable 100rel at this time.
- -->
- <!--<param name="enable-100rel" value="true"/>-->
-
- <!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
- <!-- RFC3263 Section 4.3 -->
- <!--<param name="disable-srv503" value="true"/>-->
-
- <!-- Enable Compact SIP headers. -->
- <!--<param name="enable-compact-headers" value="true"/>-->
- <!--
- enable/disable session timers
- -->
- <!--<param name="enable-timer" value="false"/>-->
- <!--<param name="minimum-session-expires" value="120"/>-->
- <param name="apply-inbound-acl" value="domains"/>
- <!--
- This defines your local network, by default we detect your local network
- and create this localnet.auto ACL for this.
- -->
- <param name="local-network-acl" value="localnet.auto"/>
- <!--<param name="apply-register-acl" value="domains"/>-->
- <!--<param name="dtmf-type" value="info"/>-->
-
-
- <!-- 'true' means every time 'first-only' means on the first register -->
- <!--<param name="send-message-query-on-register" value="true"/>-->
-
- <!-- 'true' means every time 'first-only' means on the first register -->
- <!--<param name="send-presence-on-register" value="first-only"/> -->
-
-
- <!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
- <!-- Remote-Party-ID header -->
- <!--<param name="caller-id-type" value="rpid"/>-->
-
- <!-- P-*-Identity family of headers -->
- <!--<param name="caller-id-type" value="pid"/>-->
-
- <!-- neither one -->
- <!--<param name="caller-id-type" value="none"/>-->
-
-
-
- <param name="record-path" value="$${recordings_dir}"/>
- <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
- <!--enable to use presence -->
- <param name="manage-presence" value="true"/>
- <!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info -->
- <!--<param name="presence-probe-on-register" value="true"/>-->
- <!--<param name="manage-shared-appearance" value="true"/>-->
- <!-- used to share presence info across sofia profiles -->
- <!-- Name of the db to use for this profile -->
- <!--<param name="dbname" value="share_presence"/>-->
- <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
- <param name="presence-privacy" value="$${presence_privacy}"/>
- <!-- ************************************************* -->
-
- <!-- This setting is for AAL2 bitpacking on G726 -->
- <!-- <param name="bitpacking" value="aal2"/> -->
- <!--max number of open dialogs in proceeding -->
- <!--<param name="max-proceeding" value="1000"/>-->
- <!--session timers for all call to expire after the specified seconds -->
- <!--<param name="session-timeout" value="1800"/>-->
- <!-- Can be 'true' or 'contact' -->
- <!--<param name="multiple-registrations" value="contact"/>-->
- <!--set to 'greedy' if you want your codec list to take precedence -->
- <param name="inbound-codec-negotiation" value="generous"/>
- <!-- if you want to send any special bind params of your own -->
- <!--<param name="bind-params" value="transport=udp"/>-->
- <!--<param name="unregister-on-options-fail" value="true"/>-->
-
- <!-- TLS: disabled by default, set to "true" to enable -->
- <param name="tls" value="$${internal_ssl_enable}"/>
- <!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
- <param name="tls-only" value="false"/>
- <!-- additional bind parameters for TLS -->
- <param name="tls-bind-params" value="transport=tls"/>
- <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
- <param name="tls-sip-port" value="$${internal_tls_port}"/>
- <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
- <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
- <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
- <param name="tls-passphrase" value=""/>
- <!-- Verify the date on TLS certificates -->
- <param name="tls-verify-date" value="true"/>
- <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
- <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
- <param name="tls-verify-policy" value="none"/>
- <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
- <param name="tls-verify-depth" value="2"/>
- <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
- <param name="tls-verify-in-subjects" value=""/>
- <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
- <param name="tls-version" value="$${sip_tls_version}"/>
-
- <!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
- (reduces delay on latent connections default true, must be disabled explicitly)-->
- <!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
-
- <!--If you don't want to pass through timestamps from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
- <!--<param name="rtp-rewrite-timestamps" value="true"/>-->
- <!--<param name="pass-rfc2833" value="true"/>-->
- <!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
- <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
-
- <!--Uncomment to set all inbound calls to no media mode-->
- <!--<param name="inbound-bypass-media" value="true"/>-->
-
- <!--Uncomment to set all inbound calls to proxy media mode-->
- <!--<param name="inbound-proxy-media" value="true"/>-->
-
- <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
- <!--<param name="inbound-late-negotiation" value="true"/>-->
-
- <!-- this lets anything register -->
- <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
- <!-- <param name="accept-blind-reg" value="true"/> -->
-
- <!-- accept any authentication without actually checking (not a good feature for most people) -->
- <!-- <param name="accept-blind-auth" value="true"/> -->
-
- <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
- <!-- <param name="suppress-cng" value="true"/> -->
-
- <!--TTL for nonce in sip auth-->
- <param name="nonce-ttl" value="60"/>
- <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
- that the originator is using-->
- <!--<param name="disable-transcoding" value="true"/>-->
- <!-- Handle 302 Redirect in the dialplan -->
- <!--<param name="manual-redirect" value="true"/> -->
- <!-- Disable Transfer -->
- <!--<param name="disable-transfer" value="true"/> -->
- <!-- Disable Register -->
- <!--<param name="disable-register" value="true"/> -->
- <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
- <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
- <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
- <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
- <param name="auth-calls" value="$${internal_auth_calls}"/>
- <!-- Force the user and auth-user to match. -->
- <param name="inbound-reg-force-matching-username" value="true"/>
- <!-- on authed calls, authenticate *all* the packets not just invite -->
- <param name="auth-all-packets" value="false"/>
-
- <!-- external_sip_ip
- Used as the public IP address for SDP.
- Can be an one of:
- ip address - "12.34.56.78"
- a stun server lookup - "stun:stun.server.com"
- a DNS name - "host:host.server.com"
- auto - Use guessed ip.
- auto-nat - Use ip learned from NAT-PMP or UPNP
- -->
- <param name="ext-rtp-ip" value="auto-nat"/>
- <param name="ext-sip-ip" value="auto-nat"/>
-
- <!-- rtp inactivity timeout -->
- <param name="rtp-timeout-sec" value="300"/>
- <param name="rtp-hold-timeout-sec" value="1800"/>
- <!-- VAD choose one (out is a good choice); -->
- <!-- <param name="vad" value="in"/> -->
- <!-- <param name="vad" value="out"/> -->
- <!-- <param name="vad" value="both"/> -->
- <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
- <!--
- These are enabled to make the default config work better out of the box.
- If you need more than ONE domain you'll need to not use these options.
-
- -->
- <!--all inbound reg will look in this domain for the users -->
- <param name="force-register-domain" value="$${domain}"/>
- <!--force the domain in subscriptions to this value -->
- <param name="force-subscription-domain" value="$${domain}"/>
- <!--all inbound reg will stored in the db using this domain -->
- <param name="force-register-db-domain" value="$${domain}"/>
-
- <!--<param name="delete-subs-on-register" value="false"/>-->
-
- <!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call-->
- <!--<param name="rtcp-audio-interval-msec" value="5000"/>-->
- <!--<param name="rtcp-video-interval-msec" value="5000"/>-->
-
- <!--force suscription expires to a lower value than requested-->
- <!--<param name="force-subscription-expires" value="60"/>-->
- <!-- disable register and transfer which may be undesirable in a public switch -->
- <!--<param name="disable-transfer" value="true"/>-->
- <!--<param name="disable-register" value="true"/>-->
-
- <!--
- enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
- right away, proxy waits until the call has been answered then sends accepts
- -->
- <!--<param name="enable-3pcc" value="true"/>-->
-
- <!-- use at your own risk or if you know what this does.-->
- <!--<param name="NDLB-force-rport" value="true"/>-->
- <!--
- Choose the realm challenge key. Default is auto_to if not set.
-
- auto_from - uses the from field as the value for the sip realm.
- auto_to - uses the to field as the value for the sip realm.
- <anyvalue> - you can input any value to use for the sip realm.
-
- If you want URL dialing to work you'll want to set this to auto_from.
-
- If you use any other value besides auto_to or auto_from you'll loose
- the ability to do multiple domains.
-
- Note: comment out to restore the behavior before 2008-09-29
-
- -->
- <param name="challenge-realm" value="auto_from"/>
- <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
- <!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
- <!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
- <!-- on outbound calls set the callid to match the uuid of the session -->
- <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
- <!-- set to false disable this feature -->
- <!--<param name="rtp-autofix-timing" value="false"/>-->
-
- <!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
- <!--<param name="pass-callee-id" value="false"/>-->
-
- <!-- clear clears them all or supply the name to add or the name prefixed with ~ to remove
- valid values:
-
- clear
- CISCO_SKIP_MARK_BIT_2833
- SONUS_SEND_INVALID_TIMESTAMP_2833
-
- -->
- <!--<param name="auto-rtp-bugs" data="clear"/>-->
-
- <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
- <!--<param name="disable-srv" value="false" />-->
- <!--<param name="disable-naptr" value="false" />-->
-
- <!-- The following can be used to fine-tune timers within sofia's transport layer
- Those settings are for advanced users and can safely be left as-is -->
-
- <!-- Initial retransmission interval (in milliseconds).
- Set the T1 retransmission interval used by the SIP transaction engine.
- The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. -->
- <!-- <param name="timer-T1" value="500" /> -->
-
- <!-- Transaction timeout (defaults to T1 * 64).
- Set the T1x64 timeout value used by the SIP transaction engine.
- The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine.
- The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
- <!-- <param name="timer-T1X64" value="32000" /> -->
-
-
- <!-- Maximum retransmission interval (in milliseconds).
- Set the maximum retransmission interval used by the SIP transaction engine.
- The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine.
- Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially
- until the timer B fires. -->
- <!-- <param name="timer-T2" value="4000" /> -->
-
- <!--
- Transaction lifetime (in milliseconds).
- Set the lifetime for completed transactions used by the SIP transaction engine.
- A completed transaction is kept around for the duration of T4 in order to catch late responses.
- The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
- <!-- <param name="timer-T4" value="4000" /> -->
-
- <!-- Turn on a jitterbuffer for every call -->
- <!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
-
-
- <!-- By default mod_sofia will ignore the codecs in the sdp for hold/unhold operations
- Set this to true if you want to actually parse the sdp and re-negotiate the codec during hold/unhold.
- It's probably not what you want so stick with the default unless you really need to change this.
- -->
- <!--<param name="renegotiate-codec-on-hold" value="true"/>-->
-
- </settings>
-</profile>
-
+++ /dev/null
-<include>
- <!--<gateway name="asterlink.com">-->
- <!--/// account username *required* ///-->
- <!--<param name="username" value="cluecon"/>-->
- <!--/// auth realm: *optional* same as gateway name, if blank ///-->
- <!--<param name="realm" value="asterlink.com"/>-->
- <!--/// username to use in from: *optional* same as username, if blank ///-->
- <!--<param name="from-user" value="cluecon"/>-->
- <!--/// domain to use in from: *optional* same as realm, if blank ///-->
- <!--<param name="from-domain" value="asterlink.com"/>-->
- <!--/// account password *required* ///-->
- <!--<param name="password" value="2007"/>-->
- <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
- <!--<param name="extension" value="cluecon"/>-->
- <!--/// proxy host: *optional* same as realm, if blank ///-->
- <!--<param name="proxy" value="asterlink.com"/>-->
- <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
- <!--<param name="register-proxy" value="mysbc.com"/>-->
- <!--/// expire in seconds: *optional* 3600, if blank ///-->
- <!--<param name="expire-seconds" value="60"/>-->
- <!--/// do not register ///-->
- <!--<param name="register" value="false"/>-->
- <!-- which transport to use for register -->
- <!--<param name="register-transport" value="udp"/>-->
- <!--How many seconds before a retry when a failure or timeout occurs -->
- <!--<param name="retry-seconds" value="30"/>-->
- <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
- <!--<param name="caller-id-in-from" value="false"/>-->
- <!--extra sip params to send in the contact-->
- <!--<param name="contact-params" value="tport=tcp"/>-->
- <!-- Put the extension in the contact -->
- <!--<param name="extension-in-contact" value="true"/>-->
- <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
- <!--<param name="ping" value="25"/>-->
- <!--<param name="cid-type" value="rpid"/>-->
- <!--</gateway>-->
-</include>
--- /dev/null
+<configuration name="sofia.conf" description="sofia endpoint">
+ <global_settings>
+ <param name="log-level" value="0"/>
+ <param name="auto-restart" value="false"/>
+ <param name="debug-presence" value="0"/>
+ <!-- <param name="capture-server" value="udp:homer.example.com:5060"/> -->
+ </global_settings>
+ <profiles>
+ <!--
+ This is a sofia sip profile/user agent. This will service exactly one
+ ip and port. In FreeSWITCH you can run multiple sip user agents on
+ their own ip and port.
+ -->
+ <profile name="example">
+ <gateways>
+ <gateway name="example-gateway">
+ <!-- account username (required) -->
+ <param name="username" value="cluecon"/>
+ <!-- auth realm (same as gateway name, if blank) -->
+ <param name="realm" value="example.com"/>
+ <!-- username to use in from (same as username, if blank) -->
+ <param name="from-user" value="cluecon"/>
+ <!-- domain to use in from (same as realm, if blank) /// -->
+ <param name="from-domain" value="example.com"/>
+ <!-- account password (required) -->
+ <param name="password" value="xxxx"/>
+ <!-- extension for inbound calls (same as username, if blank) -->
+ <param name="extension" value="cluecon"/>
+ <!-- proxy host (same as realm, if blank) -->
+ <param name="proxy" value="example.com"/>
+ <!-- send register to this proxy (same as proxy, if blank) -->
+ <param name="register-proxy" value="example.com"/>
+ <!-- expire in seconds (3600, if blank) -->
+ <param name="expire-seconds" value="600"/>
+ <!-- do not register -->
+ <param name="register" value="false"/>
+ <!-- which transport to use for register -->
+ <param name="register-transport" value="tcp"/>
+ <!-- how many seconds before a retry when a failure or timeout occurs
+ -->
+ <param name="retry-seconds" value="30"/>
+ <!-- use the callerid of an inbound call in the from field on outbound
+ calls via this gateway -->
+ <param name="caller-id-in-from" value="false"/>
+ <!-- extra sip params to send in the contact -->
+ <param name="contact-params" value="tport=tcp"/>
+ <!-- put the extension in the contact -->
+ <param name="extension-in-contact" value="true"/>
+ <!-- send an options ping every x seconds, failure will unregister
+ and/or mark it down -->
+ <param name="ping" value="25"/>
+ <!-- callerid header mechanism -->
+ <param name="cid-type" value="rpid"/>
+ </gateway>
+ </gateways>
+ <aliases>
+ <!-- aliases are other names that will work as a valid profile name for
+ this profile -->
+ <alias name="default"/>
+ </aliases>
+ <domains>
+ <!-- indicator to parse the directory for domains with parse="true" to
+ get gateways -->
+ <!-- <domain name="$${domain}" parse="true"/> -->
+ <!-- indicator to parse the directory for domains with parse="true" to
+ get gateways and alias every domain to this profile -->
+ <!-- <domain name="all" alias="true" parse="true"/> -->
+ <domain name="all" alias="true" parse="false"/>
+ </domains>
+ <settings>
+ <!-- When calls are in no media this will bring them back to media when
+ you press the hold button. -->
+ <!-- <param name="media-option" value="resume-media-on-hold"/> -->
+ <!-- This will allow a call after an attended transfer go back to bypass
+ media after an attended transfer. -->
+ <!-- <param name="media-option" value="bypass-media-after-att-xfer"/> -->
+ <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
+ <param name="debug" value="0"/>
+ <!-- If you want FreeSWITCH to shutdown if this profile fails to load,
+ uncomment the next line. -->
+ <!-- <param name="shutdown-on-fail" value="true"/> -->
+ <param name="sip-trace" value="no"/>
+ <param name="sip-capture" value="no"/>
+
+ <!-- Use presence_map.conf.xml to convert extension regex to presence
+ protos for routing -->
+ <!-- <param name="presence-proto-lookup" value="true"/> -->
+
+
+ <!-- Don't be picky about negotiated DTMF just always offer 2833 and
+ accept both 2833 and INFO -->
+ <!-- <param name="liberal-dtmf" value="true"/> -->
+
+ <!--
+ Sometimes, in extremely rare edge cases, the Sofia SIP stack may
+ stop responding. These options allow you to enable and control a
+ watchdog on the Sofia SIP stack so that if it stops responding for
+ the specified number of milliseconds, it will cause FreeSWITCH to
+ crash immediately. This is useful if you run in an HA environment
+ and need to ensure automated recovery from such a condition. Note
+ that if your server is idle a lot, the watchdog may fire due to not
+ receiving any SIP messages. Thus, if you expect your system to be
+ idle, you should leave the watchdog disabled. It can be toggled on
+ and off through the FreeSWITCH CLI either on an individual profile
+ basis or globally for all profiles. So, if you run in an HA
+ environment with a master and slave, you should use the CLI to make
+ sure the watchdog is only enabled on the master.
+
+ If such crash occurs, FreeSWITCH will dump core if allowed. The
+ stacktrace will include function watchdog_triggered_abort().
+ -->
+ <param name="watchdog-enabled" value="no"/>
+ <param name="watchdog-step-timeout" value="30000"/>
+ <param name="watchdog-event-timeout" value="30000"/>
+
+ <param name="log-auth-failures" value="false"/>
+ <param name="forward-unsolicited-mwi-notify" value="false"/>
+
+ <param name="context" value="public"/>
+ <param name="rfc2833-pt" value="101"/>
+ <!-- port to bind to for sip traffic -->
+ <param name="sip-port" value="$${internal_sip_port}"/>
+ <param name="dialplan" value="XML"/>
+ <param name="dtmf-duration" value="2000"/>
+ <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
+ <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
+ <param name="rtp-timer-name" value="soft"/>
+ <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
+ <param name="rtp-ip" value="$${local_ip_v4}"/>
+ <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
+ <param name="sip-ip" value="$${local_ip_v4}"/>
+ <param name="hold-music" value="$${hold_music}"/>
+ <param name="apply-nat-acl" value="nat.auto"/>
+
+ <!-- (default true) set to false if you do not wish to have called party
+ info in 1XX responses -->
+ <!-- <param name="cid-in-1xx" value="false"/> -->
+
+ <!-- extended info parsing -->
+ <!-- <param name="extended-info-parsing" value="true"/> -->
+
+ <!-- <param name="aggressive-nat-detection" value="true"/> -->
+ <!-- There are known issues (asserts and segfaults) when 100rel is
+ enabled. It is not recommended to enable 100rel at this time. -->
+ <!-- <param name="enable-100rel" value="true"/> -->
+
+ <!-- uncomment if you don't wish to try a next SRV destination on 503
+ response -->
+ <!-- RFC3263 Section 4.3 -->
+ <!-- <param name="disable-srv503" value="true"/> -->
+
+ <!-- Enable Compact SIP headers. -->
+ <!-- <param name="enable-compact-headers" value="true"/> -->
+ <!-- enable/disable session timers -->
+ <!-- <param name="enable-timer" value="false"/> -->
+ <!-- <param name="minimum-session-expires" value="120"/> -->
+ <param name="apply-inbound-acl" value="domains"/>
+ <!-- This defines your local network, by default we detect your local
+ network and create this localnet.auto ACL for this. -->
+ <param name="local-network-acl" value="localnet.auto"/>
+ <!-- <param name="apply-register-acl" value="domains"/> -->
+ <!-- <param name="dtmf-type" value="info"/> -->
+
+ <!-- 'true' means every time 'first-only' means on the first register -->
+ <!-- <param name="send-message-query-on-register" value="true"/> -->
+
+ <!-- 'true' means every time 'first-only' means on the first register -->
+ <!-- <param name="send-presence-on-register" value="first-only"/> -->
+
+ <!-- Caller-ID type (choose one, can be overridden by inbound call type
+ and/or sip_cid_type channel variable -->
+ <!-- Remote-Party-ID header -->
+ <!-- <param name="caller-id-type" value="rpid"/> -->
+
+ <!-- P-*-Identity family of headers -->
+ <!-- <param name="caller-id-type" value="pid"/> -->
+
+ <!-- neither one -->
+ <!-- <param name="caller-id-type" value="none"/> -->
+
+ <param name="record-path" value="$${recordings_dir}"/>
+ <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
+ <!-- enable to use presence -->
+ <param name="manage-presence" value="true"/>
+ <!-- send a presence probe on each register to query devices to send
+ presence instead of sending presence with less info -->
+ <!-- <param name="presence-probe-on-register" value="true"/> -->
+ <!-- <param name="manage-shared-appearance" value="true"/> -->
+ <!-- used to share presence info across sofia profiles -->
+ <!-- Name of the db to use for this profile -->
+ <!-- <param name="dbname" value="share_presence"/> -->
+ <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
+ <param name="presence-privacy" value="$${presence_privacy}"/>
+
+ <!-- This setting is for AAL2 bitpacking on G726 -->
+ <!-- <param name="bitpacking" value="aal2"/> -->
+ <!-- max number of open dialogs in proceeding -->
+ <!-- <param name="max-proceeding" value="1000"/> -->
+ <!-- session timers for all call to expire after the specified seconds -->
+ <!-- <param name="session-timeout" value="1800"/> -->
+ <!-- Can be 'true' or 'contact' -->
+ <!-- <param name="multiple-registrations" value="contact"/> -->
+ <!-- set to 'greedy' if you want your codec list to take precedence -->
+ <param name="inbound-codec-negotiation" value="generous"/>
+ <!-- if you want to send any special bind params of your own -->
+ <!-- <param name="bind-params" value="transport=udp"/> -->
+ <!-- <param name="unregister-on-options-fail" value="true"/> -->
+
+ <!-- TLS: disabled by default, set to "true" to enable -->
+ <param name="tls" value="$${internal_ssl_enable}"/>
+ <!-- Set to true to not bind on the normal sip-port but only on the TLS
+ port -->
+ <param name="tls-only" value="false"/>
+ <!-- additional bind parameters for TLS -->
+ <param name="tls-bind-params" value="transport=tls"/>
+ <!-- Port to listen on for TLS requests. (5061 will be used if
+ unspecified) -->
+ <param name="tls-sip-port" value="$${internal_tls_port}"/>
+ <!-- Location of the agent.pem and cafile.pem ssl certificates (needed
+ for TLS server) -->
+ <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
+ <!-- Optionally set the passphrase password used by openSSL to
+ encrypt/decrypt TLS private key files -->
+ <param name="tls-passphrase" value=""/>
+ <!-- Verify the date on TLS certificates -->
+ <param name="tls-verify-date" value="true"/>
+ <!-- TLS verify policy, when registering/inviting gateways with other
+ servers (outbound) or handling inbound registration/invite requests
+ how should we verify their certificate -->
+ <!-- set to 'in' to only verify incoming connections, 'out' to only
+ verify outgoing connections, 'all' to verify all connections, also
+ 'in_subjects', 'out_subjects' and 'all_subjects' for subject
+ validation. Multiple policies can be split with a '|' pipe -->
+ <param name="tls-verify-policy" value="none"/>
+ <!-- Certificate max verify depth to use for validating peer TLS
+ certificates when the verify policy is not none -->
+ <param name="tls-verify-depth" value="2"/>
+ <!-- If the tls-verify-policy is set to subjects_all or subjects_in this
+ sets which subjects are allowed, multiple subjects can be split
+ with a '|' pipe -->
+ <param name="tls-verify-in-subjects" value=""/>
+ <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not
+ work with TLSv1 -->
+ <param name="tls-version" value="$${sip_tls_version}"/>
+
+ <!-- turn on auto-flush during bridge (skip timer sleep when the socket
+ already has data) (reduces delay on latent connections default
+ true, must be disabled explicitly) -->
+ <!-- <param name="rtp-autoflush-during-bridge" value="false"/> -->
+
+ <!-- If you don't want to pass through timestamps from 1 RTP call to
+ another (on a per call basis with rtp_rewrite_timestamps chanvar)
+ -->
+ <!-- <param name="rtp-rewrite-timestamps" value="true"/> -->
+ <!-- <param name="pass-rfc2833" value="true"/> -->
+ <!-- If you have ODBC support and a working dsn you can use it instead
+ of SQLite -->
+ <!-- <param name="odbc-dsn" value="dsn:user:pass"/> -->
+
+ <!-- Uncomment to set all inbound calls to no media mode -->
+ <!-- <param name="inbound-bypass-media" value="true"/> -->
+
+ <!-- Uncomment to set all inbound calls to proxy media mode -->
+ <!-- <param name="inbound-proxy-media" value="true"/> -->
+
+ <!-- Uncomment to let calls hit the dialplan *before* you decide if the
+ codec is ok -->
+ <!-- <param name="inbound-late-negotiation" value="true"/> -->
+
+ <!-- this lets anything register -->
+ <!-- comment the next line and uncomment one or both of the other 2
+ lines for call authentication -->
+ <!-- <param name="accept-blind-reg" value="true"/> -->
+
+ <!-- accept any authentication without actually checking (not a good
+ feature for most people) -->
+ <!-- <param name="accept-blind-auth" value="true"/> -->
+
+ <!-- suppress CNG on this profile or per call with the 'suppress_cng'
+ variable -->
+ <!-- <param name="suppress-cng" value="true"/> -->
+
+ <!-- TTL for nonce in sip auth -->
+ <param name="nonce-ttl" value="60"/>
+ <!-- Uncomment if you want to force the outbound leg of a bridge to only
+ offer the codec that the originator is using -->
+ <!-- <param name="disable-transcoding" value="true"/> -->
+ <!-- Handle 302 Redirect in the dialplan -->
+ <!-- <param name="manual-redirect" value="true"/> -->
+ <!-- Disable Transfer -->
+ <!-- <param name="disable-transfer" value="true"/> -->
+ <!-- Disable Register -->
+ <!-- <param name="disable-register" value="true"/> -->
+ <!-- Used for when phones respond to a challenged ACK with method INVITE
+ in the hash -->
+ <!-- <param name="NDLB-broken-auth-hash" value="true"/> -->
+ <!-- add a ;received="<ip>:<port>" to the contact when replying to
+ register for nat handling -->
+ <!-- <param name="NDLB-received-in-nat-reg-contact" value="true"/> -->
+ <param name="auth-calls" value="$${internal_auth_calls}"/>
+ <!-- Force the user and auth-user to match. -->
+ <param name="inbound-reg-force-matching-username" value="true"/>
+ <!-- on authed calls, authenticate *all* the packets not just invite -->
+ <param name="auth-all-packets" value="false"/>
+
+ <!-- external_sip_ip
+ Used as the public IP address for SDP.
+ Can be an one of:
+ ip address - "12.34.56.78"
+ a stun server lookup - "stun:stun.server.com"
+ a DNS name - "host:host.server.com"
+ auto - Use guessed ip.
+ auto-nat - Use ip learned from NAT-PMP or UPNP
+ -->
+ <param name="ext-rtp-ip" value="auto-nat"/>
+ <param name="ext-sip-ip" value="auto-nat"/>
+
+ <!-- rtp inactivity timeout -->
+ <param name="rtp-timeout-sec" value="300"/>
+ <param name="rtp-hold-timeout-sec" value="1800"/>
+ <!-- VAD choose one (out is a good choice); -->
+ <!-- <param name="vad" value="in"/> -->
+ <!-- <param name="vad" value="out"/> -->
+ <!-- <param name="vad" value="both"/> -->
+ <!-- <param name="alias" value="sip:10.0.1.251:5555"/> -->
+ <!--
+ These are enabled to make the default config work better out of the
+ box. If you need more than ONE domain you'll need to not use these
+ options.
+ -->
+ <!-- all inbound reg will look in this domain for the users -->
+ <param name="force-register-domain" value="$${domain}"/>
+ <!-- force the domain in subscriptions to this value -->
+ <param name="force-subscription-domain" value="$${domain}"/>
+ <!-- all inbound reg will stored in the db using this domain -->
+ <param name="force-register-db-domain" value="$${domain}"/>
+
+ <!-- <param name="delete-subs-on-register" value="false"/> -->
+
+ <!-- enable rtcp on every channel also can be done per leg basis with
+ rtcp_audio_interval_msec variable set to passthru to pass it across
+ a call -->
+ <!-- <param name="rtcp-audio-interval-msec" value="5000"/> -->
+ <!-- <param name="rtcp-video-interval-msec" value="5000"/> -->
+
+ <!-- force suscription expires to a lower value than requested -->
+ <!-- <param name="force-subscription-expires" value="60"/> -->
+ <!-- disable register and transfer which may be undesirable in a public
+ switch -->
+ <!-- <param name="disable-transfer" value="true"/> -->
+ <!-- <param name="disable-register" value="true"/> -->
+
+ <!--
+ enable-3pcc can be set to either 'true' or 'proxy', true accepts
+ the call right away, proxy waits until the call has been answered
+ then sends accepts
+ -->
+ <!-- <param name="enable-3pcc" value="true"/> -->
+
+ <!-- use at your own risk or if you know what this does. -->
+ <!-- <param name="NDLB-force-rport" value="true"/> -->
+ <!--
+ Choose the realm challenge key. Default is auto_to if not set.
+
+ auto_from - uses the from field as the value for the sip realm.
+ auto_to - uses the to field as the value for the sip realm.
+ <anyvalue> - you can input any value to use for the sip realm.
+
+ If you want URL dialing to work you'll want to set this to auto_from.
+
+ If you use any other value besides auto_to or auto_from you'll loose
+ the ability to do multiple domains.
+
+ Note: comment out to restore the behavior before 2008-09-29
+
+ -->
+ <param name="challenge-realm" value="auto_from"/>
+ <!-- <param name="disable-rtp-auto-adjust" value="true"/> -->
+ <!-- on inbound calls make the uuid of the session equal to the sip call
+ id of that call -->
+ <!-- <param name="inbound-use-callid-as-uuid" value="true"/> -->
+ <!-- on outbound calls set the callid to match the uuid of the session
+ -->
+ <!-- <param name="outbound-use-uuid-as-callid" value="true"/> -->
+ <!-- set to false disable this feature -->
+ <!-- <param name="rtp-autofix-timing" value="false"/> -->
+
+ <!-- set this param to false if your gateway for some reason hates X-
+ headers that it is supposed to ignore -->
+ <!-- <param name="pass-callee-id" value="false"/> -->
+
+ <!-- clear clears them all or supply the name to add or the name
+ prefixed with ~ to remove valid values:
+
+ clear
+ CISCO_SKIP_MARK_BIT_2833
+ SONUS_SEND_INVALID_TIMESTAMP_2833
+
+ -->
+ <!-- <param name="auto-rtp-bugs" data="clear"/> -->
+
+ <!-- the following can be used as workaround with bogus SRV/NAPTR
+ records -->
+ <!-- <param name="disable-srv" value="false" /> -->
+ <!-- <param name="disable-naptr" value="false" /> -->
+
+ <!-- The following can be used to fine-tune timers within sofia's
+ transport layer Those settings are for advanced users and can
+ safely be left as-is -->
+
+ <!-- Initial retransmission interval (in milliseconds).
+
+ Set the T1 retransmission interval used by the SIP transaction
+ engine.
+
+ The T1 is the initial duration used by request retransmission
+ timers A and E (UDP) as well as response retransmission timer G.
+ -->
+ <!-- <param name="timer-T1" value="500" /> -->
+
+ <!-- Transaction timeout (defaults to T1 * 64).
+
+ Set the T1x64 timeout value used by the SIP transaction engine.
+
+ The T1x64 is duration used for timers B, F, H, and J (UDP) by the
+ SIP transaction engine.
+
+ The timeout value T1x64 can be adjusted separately from the initial
+ retransmission interval T1. -->
+ <!-- <param name="timer-T1X64" value="32000" /> -->
+
+
+ <!-- Maximum retransmission interval (in milliseconds).
+
+ Set the maximum retransmission interval used by the SIP transaction
+ engine.
+
+ The T2 is the maximum duration used for the timers E (UDP) and G by
+ the SIP transaction engine.
+
+ Note that the timer A is not capped by T2. Retransmission interval
+ of INVITE requests grows exponentially until the timer B fires.
+ -->
+ <!-- <param name="timer-T2" value="4000" /> -->
+
+ <!--
+ Transaction lifetime (in milliseconds).
+
+ Set the lifetime for completed transactions used by the SIP
+ transaction engine.
+
+ A completed transaction is kept around for the duration of T4 in
+ order to catch late responses.
+
+ The T4 is the maximum duration for the messages to stay in the
+ network and the duration of SIP timer K. -->
+ <!-- <param name="timer-T4" value="4000" /> -->
+
+ <!-- Turn on a jitterbuffer for every call -->
+ <!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
+
+
+ <!-- By default mod_sofia will ignore the codecs in the sdp for
+ hold/unhold operations Set this to true if you want to actually
+ parse the sdp and re-negotiate the codec during hold/unhold. It's
+ probably not what you want so stick with the default unless you
+ really need to change this.
+ -->
+ <!-- <param name="renegotiate-codec-on-hold" value="true"/> -->
+ </settings>
+ </profile>
+ </profiles>
+</configuration>
projects / thirdparty / freeswitch.git / commitdiff
