-<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
+<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="external">
<!-- This profile is only for outbound registrations to providers -->
<gateways>
<!-- This could be set to "passive" -->
<param name="manage-presence" value="passive"/>
- <!-- 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.
+ <!-- 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="$${domain}"/>
<param name="auth-calls" value="false"/>
<param name="rtp-timeout-sec" value="1800"/>
<!--
- DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
+ DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
-->
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
<!--<param name="from-domain" value="asterlink.com"/>-->
<!--/// account password *required* ///-->
- <!--<param name="password" value="2007"/>-->
+ <!--<param name="password" value="2007"/>-->
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
<!--<param name="extension" value="cluecon"/>-->
<!--/// proxy host: *optional* same as realm, if blank ///-->
<param name="tls-cert-dir" value="$${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-->
+ <!--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"/>-->
</settings>
</profile>
-
<!--
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.
-->
<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"/>
+ <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.
+ 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.
+ 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="dtmf-type" value="info"/>-->
<param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
-
+
<!-- This setting is for AAL2 bitpacking on G726 -->
<!-- <param name="bitpacking" value="aal2"/> -->
<!--max number of open dialogs in proceeding -->
<param name="tls-cert-dir" value="$${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 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-->
+ <!--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"/>-->
<!-- <param name="vad" value="out"/> -->
<!-- <param name="vad" value="both"/> -->
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
-
+
<!--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="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 -->
+ <!-- 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
+ 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="outbound-use-uuid-as-callid" value="true"/>-->
</settings>
</profile>
-
<!--/// 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"/>-->
+ <!--<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 ///-->
-<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
+<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="external">
<!-- This profile is only for outbound registrations to providers -->
<gateways>
<aliases>
<alias name="outbound"/>
- <alias name="nat"/> <!-- for backwards compatibility -->
+ <alias name="nat"/> <!-- for backwards compatibility -->
</aliases>
<domains>
<!-- This could be set to "passive" -->
<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.
+ <!-- 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="auth-calls" value="false"/>
<param name="rtp-timeout-sec" value="1800"/>
<!--
- DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
+ 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}"/>
<!--/// 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"/>-->
+ <!--<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="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-->
+ <!--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"/>-->
</settings>
</profile>
-
<!--
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.
-->
<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="$${domain}" parse="true"/>-->
<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
<!--<domain name="all" alias="true" parse="true"/>-->
- <domain name="all" alias="true" parse="false"/>
+ <domain name="all" alias="true" parse="false"/>
</domains>
-
+
<settings>
<!--
- When calls are in no media this will bring them back to media
- when you press the hold button.
+ When calls are in no media this will bring them back to media
+ when you press the hold button.
-->
<!--<param name="media-option" value="resume-media-on-hold"/> -->
<!--
- This will allow a call after an attended transfer go back to
- bypass media after an attended transfer.
+ This will allow a call after an attended transfer go back to
+ bypass media after an attended transfer.
-->
<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
<!--<param name="dbname" value="share_presence"/>-->
<!--<param name="presence-hosts" value="$${domain}"/>-->
<!-- ************************************************* -->
-
+
<!-- This setting is for AAL2 bitpacking on G726 -->
<!-- <param name="bitpacking" value="aal2"/> -->
<!--max number of open dialogs in proceeding -->
<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 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-->
+ <!--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"/>-->
<!--<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 -->
+ <!-- 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
+ 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="outbound-use-uuid-as-callid" value="true"/>-->
</settings>
</profile>
-
<!--/// 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"/>-->
+ <!--<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 ///-->
<profile name="external">
- <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
+ <!-- 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"/>
+ <!--
+
<alias name="outbound"/>
-->
</aliases>
<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"/> -->
+ <!-- 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="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="hold-music" value="$${hold_music}"/>
<param name="rtp-timer-name" value="soft"/>
<!--<param name="enable-100rel" value="true"/>-->
- <!--<param name="disable-srv503" value="true"/>-->
+ <!--<param name="disable-srv503" value="true"/>-->
<!-- This could be set to "passive" -->
<param name="local-network-acl" value="localnet.auto"/>
<param name="manage-presence" value="false"/>
- <!-- used to share presence info across sofia profiles
- manage-presence needs to be set to passive on this profile
- if you want it to behave as if it were the internal profile
- for presence.
+ <!-- used to share presence info across sofia profiles
+ manage-presence needs to be set to passive on this profile
+ if you want it to behave as if it were the internal profile
+ for presence.
-->
<!-- Name of the db to use for this profile -->
<!--<param name="dbname" value="share_presence"/>-->
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<!--
- DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
+ DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
-->
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="tls-verify-in-subjects" value=""/>
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<param name="tls-version" value="$${sip_tls_version}"/>
-
</settings>
</profile>
<!--/// 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"/>-->
+ <!--<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="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-->
+ <!--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"/>-->
<!-- <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.
+ 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 -->
<!-- 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" />-->
+ <!-- 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>
-
<!--
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"/>
+
<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"/>
+ <domain name="all" alias="true" parse="false"/>
</domains>
-
+
<settings>
<!-- <param name="rtp-digit-delay" value="40"/>-->
<!--
- When calls are in no media this will bring them back to media
- when you press the hold button.
+ 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.
+ 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"/> -->
+ <!-- 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
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.
+ 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="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.
+ 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"/>-->
-
+ <!-- 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
+ 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.
+ 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"/>-->
<!-- '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="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 -->
<!-- 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)-->
+ <!-- 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-->
+ <!--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"/> -->
<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
- -->
+ 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"/>
<!-- <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.
+ 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="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
+ <!--
+ 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
+ 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"/>-->
<!-- 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 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
+ 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"/>-->
-
+ <!-- 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>
-
<!--/// 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"/>-->
+ <!--<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 ///-->
projects / thirdparty / freeswitch.git / commitdiff
