whitespace cleanup
This commit is contained in:
Travis Cross
parent
9b569ec875
commit
73614127fc
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@ -1,4 +1,4 @@
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<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
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<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
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<profile name="external">
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<profile name="external">
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<!-- This profile is only for outbound registrations to providers -->
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<!-- This profile is only for outbound registrations to providers -->
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<gateways>
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<gateways>
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@ -29,10 +29,10 @@
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<!-- This could be set to "passive" -->
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<!-- This could be set to "passive" -->
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<param name="manage-presence" value="passive"/>
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<param name="manage-presence" value="passive"/>
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<!-- used to share presence info across sofia profiles
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<!-- used to share presence info across sofia profiles
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manage-presence needs to be set to passive on this profile
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manage-presence needs to be set to passive on this profile
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if you want it to behave as if it were the internal profile
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if you want it to behave as if it were the internal profile
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for presence.
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for presence.
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-->
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-->
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<!-- Name of the db to use for this profile -->
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<!-- Name of the db to use for this profile -->
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<param name="dbname" value="$${domain}"/>
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<param name="dbname" value="$${domain}"/>
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@ -48,7 +48,7 @@
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<param name="auth-calls" value="false"/>
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<param name="auth-calls" value="false"/>
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<param name="rtp-timeout-sec" value="1800"/>
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<param name="rtp-timeout-sec" value="1800"/>
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<!--
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<!--
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DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
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DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
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-->
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-->
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<param name="rtp-ip" value="$${local_ip_v4}"/>
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<param name="rtp-ip" value="$${local_ip_v4}"/>
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<param name="sip-ip" value="$${local_ip_v4}"/>
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<param name="sip-ip" value="$${local_ip_v4}"/>
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@ -9,7 +9,7 @@
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<!--/// domain to use in from: *optional* same as realm, if blank ///-->
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<!--/// domain to use in from: *optional* same as realm, if blank ///-->
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<!--<param name="from-domain" value="asterlink.com"/>-->
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<!--<param name="from-domain" value="asterlink.com"/>-->
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<!--/// account password *required* ///-->
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<!--/// account password *required* ///-->
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<!--<param name="password" value="2007"/>-->
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<!--<param name="password" value="2007"/>-->
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<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
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<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
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<!--<param name="extension" value="cluecon"/>-->
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<!--<param name="extension" value="cluecon"/>-->
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<!--/// proxy host: *optional* same as realm, if blank ///-->
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<!--/// proxy host: *optional* same as realm, if blank ///-->
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@ -53,36 +53,36 @@
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<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
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<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
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<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
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<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
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<param name="tls-version" value="$${sip_tls_version}"/>
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<param name="tls-version" value="$${sip_tls_version}"/>
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<!--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)-->
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<!--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)-->
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<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
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<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
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<!--<param name="pass-rfc2833" value="true"/>-->
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<!--<param name="pass-rfc2833" value="true"/>-->
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<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
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<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
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<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
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<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
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<!--Uncomment to set all inbound calls to no media mode-->
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<!--Uncomment to set all inbound calls to no media mode-->
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<!--<param name="inbound-bypass-media" value="true"/>-->
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<!--<param name="inbound-bypass-media" value="true"/>-->
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<!--Uncomment to set all inbound calls to proxy media mode-->
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<!--Uncomment to set all inbound calls to proxy media mode-->
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<!--<param name="inbound-proxy-media" value="true"/>-->
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<!--<param name="inbound-proxy-media" value="true"/>-->
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<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
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<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
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<!--<param name="inbound-late-negotiation" value="true"/>-->
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<!--<param name="inbound-late-negotiation" value="true"/>-->
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<!-- this lets anything register -->
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<!-- this lets anything register -->
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<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
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<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
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<!-- <param name="accept-blind-reg" value="true"/> -->
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<!-- <param name="accept-blind-reg" value="true"/> -->
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<!-- accept any authentication without actually checking (not a good feature for most people) -->
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<!-- accept any authentication without actually checking (not a good feature for most people) -->
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<!-- <param name="accept-blind-auth" value="true"/> -->
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<!-- <param name="accept-blind-auth" value="true"/> -->
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<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
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<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
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<!-- <param name="suppress-cng" value="true"/> -->
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<!-- <param name="suppress-cng" value="true"/> -->
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<!--TTL for nonce in sip auth-->
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<!--TTL for nonce in sip auth-->
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<param name="nonce-ttl" value="60"/>
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<param name="nonce-ttl" value="60"/>
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<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
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<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
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that the originator is using-->
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that the originator is using-->
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<!--<param name="disable-transcoding" value="true"/>-->
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<!--<param name="disable-transcoding" value="true"/>-->
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<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
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<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
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<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
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<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
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@ -128,4 +128,3 @@
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</settings>
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</settings>
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</profile>
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</profile>
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<!--
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<!--
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This is a sofia sip profile/user agent. This will service exactly one ip and port.
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This is a sofia sip profile/user agent. This will service exactly one ip and port.
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In FreeSWITCH you can run multiple sip user agents on their own ip and port.
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In FreeSWITCH you can run multiple sip user agents on their own ip and port.
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When you hear someone say "sofia profile" this is what they are talking about.
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When you hear someone say "sofia profile" this is what they are talking about.
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-->
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-->
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<gateways>
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<gateways>
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<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
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<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
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</gateways>
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</gateways>
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<domains>
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<domains>
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<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
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<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
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<!--<domain name="$${domain}" parse="true"/>-->
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<!--<domain name="$${domain}" parse="true"/>-->
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<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
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<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
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<!--<domain name="all" alias="true" parse="true"/>-->
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<!--<domain name="all" alias="true" parse="true"/>-->
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<domain name="all" alias="true" parse="false"/>
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<domain name="all" alias="true" parse="false"/>
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</domains>
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</domains>
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<settings>
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<settings>
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<!--
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<!--
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When calls are in no media this will bring them back to media
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When calls are in no media this will bring them back to media
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when you press the hold button.
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when you press the hold button.
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-->
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-->
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<!--<param name="media-option" value="resume-media-on-hold"/> -->
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<!--<param name="media-option" value="resume-media-on-hold"/> -->
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<!--
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<!--
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This will allow a call after an attended transfer go back to
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This will allow a call after an attended transfer go back to
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bypass media after an attended transfer.
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bypass media after an attended transfer.
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-->
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-->
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<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
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<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
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<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
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<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
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<!--<param name="dtmf-type" value="info"/>-->
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<!--<param name="dtmf-type" value="info"/>-->
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<param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
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<param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
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<!-- This setting is for AAL2 bitpacking on G726 -->
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<!-- This setting is for AAL2 bitpacking on G726 -->
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<!-- <param name="bitpacking" value="aal2"/> -->
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<!-- <param name="bitpacking" value="aal2"/> -->
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<!--max number of open dialogs in proceeding -->
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<!--max number of open dialogs in proceeding -->
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<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
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<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
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<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
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<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
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<param name="tls-version" value="$${sip_tls_version}"/>
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<param name="tls-version" value="$${sip_tls_version}"/>
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<!--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)-->
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<!--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)-->
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<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
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<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
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<!--<param name="pass-rfc2833" value="true"/>-->
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<!--<param name="pass-rfc2833" value="true"/>-->
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<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
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<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
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<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
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<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
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<!--Uncomment to set all inbound calls to no media mode-->
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<!--Uncomment to set all inbound calls to no media mode-->
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<!--<param name="inbound-bypass-media" value="true"/>-->
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<!--<param name="inbound-bypass-media" value="true"/>-->
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<!--Uncomment to set all inbound calls to proxy media mode-->
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<!--Uncomment to set all inbound calls to proxy media mode-->
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<!--<param name="inbound-proxy-media" value="true"/>-->
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<!--<param name="inbound-proxy-media" value="true"/>-->
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<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
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<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
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<!--<param name="inbound-late-negotiation" value="true"/>-->
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<!--<param name="inbound-late-negotiation" value="true"/>-->
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<!-- this lets anything register -->
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<!-- this lets anything register -->
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<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
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<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
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<!-- <param name="accept-blind-reg" value="true"/> -->
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<!-- <param name="accept-blind-reg" value="true"/> -->
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<!-- accept any authentication without actually checking (not a good feature for most people) -->
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<!-- accept any authentication without actually checking (not a good feature for most people) -->
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<!-- <param name="accept-blind-auth" value="true"/> -->
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<!-- <param name="accept-blind-auth" value="true"/> -->
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<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
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<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
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<!-- <param name="suppress-cng" value="true"/> -->
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<!-- <param name="suppress-cng" value="true"/> -->
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<!--TTL for nonce in sip auth-->
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<!--TTL for nonce in sip auth-->
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<param name="nonce-ttl" value="60"/>
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<param name="nonce-ttl" value="60"/>
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<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
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<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
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that the originator is using-->
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that the originator is using-->
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<!--<param name="disable-transcoding" value="true"/>-->
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<!--<param name="disable-transcoding" value="true"/>-->
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<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
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<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
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<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
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<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
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<!-- <param name="vad" value="out"/> -->
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<!-- <param name="vad" value="out"/> -->
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<!-- <param name="vad" value="both"/> -->
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<!-- <param name="vad" value="both"/> -->
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<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
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<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
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<!--all inbound reg will look in this domain for the users -->
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<!--all inbound reg will look in this domain for the users -->
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<param name="force-register-domain" value="$${domain}"/>
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<param name="force-register-domain" value="$${domain}"/>
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<!--all inbound reg will stored in the db using this domain -->
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<!--all inbound reg will stored in the db using this domain -->
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<!--<param name="disable-transfer" value="true"/>-->
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<!--<param name="disable-transfer" value="true"/>-->
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<!--<param name="disable-register" value="true"/>-->
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<!--<param name="disable-register" value="true"/>-->
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<!-- 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 -->
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<!-- 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 -->
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<!--<param name="enable-3pcc" value="true"/>-->
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<!--<param name="enable-3pcc" value="true"/>-->
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<!-- use at your own risk or if you know what this does.-->
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<!-- use at your own risk or if you know what this does.-->
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<!--<param name="NDLB-force-rport" value="true"/>-->
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<!--<param name="NDLB-force-rport" value="true"/>-->
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<!--
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<!--
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Choose the realm challenge key. Default is auto_to if not set.
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Choose the realm challenge key. Default is auto_to if not set.
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auto_from - uses the from field as the value for the sip realm.
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auto_to - uses the to field as the value for the sip realm.
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<anyvalue> - you can input any value to use for the sip realm.
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If you want URL dialing to work you'll want to set this to auto_from.
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auto_from - uses the from field as the value for the sip realm.
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auto_to - uses the to field as the value for the sip realm.
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If you use any other value besides auto_to or auto_from you'll loose
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<anyvalue> - you can input any value to use for the sip realm.
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the ability to do multiple domains.
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If you want URL dialing to work you'll want to set this to auto_from.
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Note: comment out to restore the behavior before 2008-09-29
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If you use any other value besides auto_to or auto_from you'll loose
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the ability to do multiple domains.
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Note: comment out to restore the behavior before 2008-09-29
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-->
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-->
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<param name="challenge-realm" value="auto_from"/>
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<param name="challenge-realm" value="auto_from"/>
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@ -186,4 +186,3 @@
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<!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
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<!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
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</settings>
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</settings>
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</profile>
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</profile>
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@ -9,7 +9,7 @@
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<!--/// domain to use in from: *optional* same as realm, if blank ///-->
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<!--/// domain to use in from: *optional* same as realm, if blank ///-->
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<!--<param name="from-domain" value="asterlink.com"/>-->
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<!--<param name="from-domain" value="asterlink.com"/>-->
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<!--/// account password *required* ///-->
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<!--/// account password *required* ///-->
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<!--<param name="password" value="2007"/>-->
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<!--<param name="password" value="2007"/>-->
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<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
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<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
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<!--<param name="extension" value="cluecon"/>-->
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<!--<param name="extension" value="cluecon"/>-->
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<!--/// proxy host: *optional* same as realm, if blank ///-->
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<!--/// proxy host: *optional* same as realm, if blank ///-->
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@ -1,4 +1,4 @@
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<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
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<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
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<profile name="external">
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<profile name="external">
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<!-- This profile is only for outbound registrations to providers -->
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<!-- This profile is only for outbound registrations to providers -->
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<gateways>
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<gateways>
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@ -7,7 +7,7 @@
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<aliases>
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<aliases>
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<alias name="outbound"/>
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<alias name="outbound"/>
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<alias name="nat"/> <!-- for backwards compatibility -->
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<alias name="nat"/> <!-- for backwards compatibility -->
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</aliases>
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</aliases>
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<domains>
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<domains>
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@ -30,10 +30,10 @@
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<!-- This could be set to "passive" -->
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<!-- This could be set to "passive" -->
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||||||
<param name="manage-presence" value="false"/>
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<param name="manage-presence" value="false"/>
|
||||||
|
|
||||||
<!-- used to share presence info across sofia profiles
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<!-- used to share presence info across sofia profiles
|
||||||
manage-presence needs to be set to passive on this profile
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manage-presence needs to be set to passive on this profile
|
||||||
if you want it to behave as if it were the internal profile
|
if you want it to behave as if it were the internal profile
|
||||||
for presence.
|
for presence.
|
||||||
-->
|
-->
|
||||||
<!-- Name of the db to use for this profile -->
|
<!-- Name of the db to use for this profile -->
|
||||||
<!--<param name="dbname" value="share_presence"/>-->
|
<!--<param name="dbname" value="share_presence"/>-->
|
||||||
|
|
@ -49,7 +49,7 @@
|
||||||
<param name="auth-calls" value="false"/>
|
<param name="auth-calls" value="false"/>
|
||||||
<param name="rtp-timeout-sec" value="1800"/>
|
<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="$${local_ip_v4}"/>
|
<param name="rtp-ip" value="$${local_ip_v4}"/>
|
||||||
<param name="sip-ip" value="$${local_ip_v4}"/>
|
<param name="sip-ip" value="$${local_ip_v4}"/>
|
||||||
|
|
|
||||||
|
|
@ -9,7 +9,7 @@
|
||||||
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
|
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
|
||||||
<!--<param name="from-domain" value="asterlink.com"/>-->
|
<!--<param name="from-domain" value="asterlink.com"/>-->
|
||||||
<!--/// account password *required* ///-->
|
<!--/// account password *required* ///-->
|
||||||
<!--<param name="password" value="2007"/>-->
|
<!--<param name="password" value="2007"/>-->
|
||||||
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
|
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
|
||||||
<!--<param name="extension" value="cluecon"/>-->
|
<!--<param name="extension" value="cluecon"/>-->
|
||||||
<!--/// proxy host: *optional* same as realm, if blank ///-->
|
<!--/// proxy host: *optional* same as realm, if blank ///-->
|
||||||
|
|
|
||||||
|
|
@ -53,36 +53,36 @@
|
||||||
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
|
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
|
||||||
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
||||||
<param name="tls-version" value="$${sip_tls_version}"/>
|
<param name="tls-version" value="$${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)-->
|
<!--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="rtp-rewrite-timestamps" value="true"/>-->
|
||||||
<!--<param name="pass-rfc2833" value="true"/>-->
|
<!--<param name="pass-rfc2833" value="true"/>-->
|
||||||
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
|
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
|
||||||
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
|
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
|
||||||
|
|
||||||
<!--Uncomment to set all inbound calls to no media mode-->
|
<!--Uncomment to set all inbound calls to no media mode-->
|
||||||
<!--<param name="inbound-bypass-media" value="true"/>-->
|
<!--<param name="inbound-bypass-media" value="true"/>-->
|
||||||
|
|
||||||
<!--Uncomment to set all inbound calls to proxy media mode-->
|
<!--Uncomment to set all inbound calls to proxy media mode-->
|
||||||
<!--<param name="inbound-proxy-media" value="true"/>-->
|
<!--<param name="inbound-proxy-media" value="true"/>-->
|
||||||
|
|
||||||
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
|
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
|
||||||
<!--<param name="inbound-late-negotiation" value="true"/>-->
|
<!--<param name="inbound-late-negotiation" value="true"/>-->
|
||||||
|
|
||||||
<!-- this lets anything register -->
|
<!-- this lets anything register -->
|
||||||
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
|
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
|
||||||
<!-- <param name="accept-blind-reg" value="true"/> -->
|
<!-- <param name="accept-blind-reg" value="true"/> -->
|
||||||
|
|
||||||
<!-- accept any authentication without actually checking (not a good feature for most people) -->
|
<!-- accept any authentication without actually checking (not a good feature for most people) -->
|
||||||
<!-- <param name="accept-blind-auth" value="true"/> -->
|
<!-- <param name="accept-blind-auth" value="true"/> -->
|
||||||
|
|
||||||
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
|
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
|
||||||
<!-- <param name="suppress-cng" value="true"/> -->
|
<!-- <param name="suppress-cng" value="true"/> -->
|
||||||
|
|
||||||
<!--TTL for nonce in sip auth-->
|
<!--TTL for nonce in sip auth-->
|
||||||
<param name="nonce-ttl" value="60"/>
|
<param name="nonce-ttl" value="60"/>
|
||||||
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
|
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
|
||||||
that the originator is using-->
|
that the originator is using-->
|
||||||
<!--<param name="disable-transcoding" value="true"/>-->
|
<!--<param name="disable-transcoding" value="true"/>-->
|
||||||
<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
|
<!-- 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="NDLB-broken-auth-hash" value="true"/>-->
|
||||||
|
|
@ -128,4 +128,3 @@
|
||||||
|
|
||||||
</settings>
|
</settings>
|
||||||
</profile>
|
</profile>
|
||||||
|
|
||||||
|
|
|
||||||
|
|
@ -1,7 +1,7 @@
|
||||||
<!--
|
<!--
|
||||||
This is a sofia sip profile/user agent. This will service exactly one ip and port.
|
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.
|
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.
|
When you hear someone say "sofia profile" this is what they are talking about.
|
||||||
-->
|
-->
|
||||||
|
|
||||||
|
|
@ -15,24 +15,24 @@
|
||||||
<gateways>
|
<gateways>
|
||||||
<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
|
<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
|
||||||
</gateways>
|
</gateways>
|
||||||
|
|
||||||
<domains>
|
<domains>
|
||||||
<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
|
<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
|
||||||
<!--<domain name="$${domain}" parse="true"/>-->
|
<!--<domain name="$${domain}" parse="true"/>-->
|
||||||
<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
|
<!-- 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="true"/>-->
|
||||||
<domain name="all" alias="true" parse="false"/>
|
<domain name="all" alias="true" parse="false"/>
|
||||||
</domains>
|
</domains>
|
||||||
|
|
||||||
<settings>
|
<settings>
|
||||||
<!--
|
<!--
|
||||||
When calls are in no media this will bring them back to media
|
When calls are in no media this will bring them back to media
|
||||||
when you press the hold button.
|
when you press the hold button.
|
||||||
-->
|
-->
|
||||||
<!--<param name="media-option" value="resume-media-on-hold"/> -->
|
<!--<param name="media-option" value="resume-media-on-hold"/> -->
|
||||||
<!--
|
<!--
|
||||||
This will allow a call after an attended transfer go back to
|
This will allow a call after an attended transfer go back to
|
||||||
bypass media after an attended transfer.
|
bypass media after an attended transfer.
|
||||||
-->
|
-->
|
||||||
<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
|
<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
|
||||||
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
|
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
|
||||||
|
|
@ -69,7 +69,7 @@
|
||||||
<!--<param name="dbname" value="share_presence"/>-->
|
<!--<param name="dbname" value="share_presence"/>-->
|
||||||
<!--<param name="presence-hosts" value="$${domain}"/>-->
|
<!--<param name="presence-hosts" value="$${domain}"/>-->
|
||||||
<!-- ************************************************* -->
|
<!-- ************************************************* -->
|
||||||
|
|
||||||
<!-- This setting is for AAL2 bitpacking on G726 -->
|
<!-- This setting is for AAL2 bitpacking on G726 -->
|
||||||
<!-- <param name="bitpacking" value="aal2"/> -->
|
<!-- <param name="bitpacking" value="aal2"/> -->
|
||||||
<!--max number of open dialogs in proceeding -->
|
<!--max number of open dialogs in proceeding -->
|
||||||
|
|
@ -94,36 +94,36 @@
|
||||||
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
|
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
|
||||||
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
||||||
<param name="tls-version" value="$${sip_tls_version}"/>
|
<param name="tls-version" value="$${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)-->
|
<!--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="rtp-rewrite-timestamps" value="true"/>-->
|
||||||
<!--<param name="pass-rfc2833" value="true"/>-->
|
<!--<param name="pass-rfc2833" value="true"/>-->
|
||||||
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
|
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
|
||||||
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
|
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
|
||||||
|
|
||||||
<!--Uncomment to set all inbound calls to no media mode-->
|
<!--Uncomment to set all inbound calls to no media mode-->
|
||||||
<!--<param name="inbound-bypass-media" value="true"/>-->
|
<!--<param name="inbound-bypass-media" value="true"/>-->
|
||||||
|
|
||||||
<!--Uncomment to set all inbound calls to proxy media mode-->
|
<!--Uncomment to set all inbound calls to proxy media mode-->
|
||||||
<!--<param name="inbound-proxy-media" value="true"/>-->
|
<!--<param name="inbound-proxy-media" value="true"/>-->
|
||||||
|
|
||||||
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
|
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
|
||||||
<!--<param name="inbound-late-negotiation" value="true"/>-->
|
<!--<param name="inbound-late-negotiation" value="true"/>-->
|
||||||
|
|
||||||
<!-- this lets anything register -->
|
<!-- this lets anything register -->
|
||||||
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
|
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
|
||||||
<!-- <param name="accept-blind-reg" value="true"/> -->
|
<!-- <param name="accept-blind-reg" value="true"/> -->
|
||||||
|
|
||||||
<!-- accept any authentication without actually checking (not a good feature for most people) -->
|
<!-- accept any authentication without actually checking (not a good feature for most people) -->
|
||||||
<!-- <param name="accept-blind-auth" value="true"/> -->
|
<!-- <param name="accept-blind-auth" value="true"/> -->
|
||||||
|
|
||||||
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
|
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
|
||||||
<!-- <param name="suppress-cng" value="true"/> -->
|
<!-- <param name="suppress-cng" value="true"/> -->
|
||||||
|
|
||||||
<!--TTL for nonce in sip auth-->
|
<!--TTL for nonce in sip auth-->
|
||||||
<param name="nonce-ttl" value="60"/>
|
<param name="nonce-ttl" value="60"/>
|
||||||
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
|
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
|
||||||
that the originator is using-->
|
that the originator is using-->
|
||||||
<!--<param name="disable-transcoding" value="true"/>-->
|
<!--<param name="disable-transcoding" value="true"/>-->
|
||||||
<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
|
<!-- 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="NDLB-broken-auth-hash" value="true"/>-->
|
||||||
|
|
@ -154,24 +154,24 @@
|
||||||
<!--<param name="disable-transfer" value="true"/>-->
|
<!--<param name="disable-transfer" value="true"/>-->
|
||||||
<!--<param name="disable-register" 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"/>-->
|
<!--<param name="enable-3pcc" value="true"/>-->
|
||||||
|
|
||||||
<!-- use at your own risk or if you know what this does.-->
|
<!-- use at your own risk or if you know what this does.-->
|
||||||
<!--<param name="NDLB-force-rport" value="true"/>-->
|
<!--<param name="NDLB-force-rport" value="true"/>-->
|
||||||
<!--
|
<!--
|
||||||
Choose the realm challenge key. Default is auto_to if not set.
|
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.
|
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.
|
||||||
If you use any other value besides auto_to or auto_from you'll loose
|
<anyvalue> - you can input any value to use for the sip realm.
|
||||||
the ability to do multiple domains.
|
|
||||||
|
If you want URL dialing to work you'll want to set this to auto_from.
|
||||||
Note: comment out to restore the behavior before 2008-09-29
|
|
||||||
|
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="challenge-realm" value="auto_from"/>
|
||||||
|
|
@ -182,4 +182,3 @@
|
||||||
<!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
|
<!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
|
||||||
</settings>
|
</settings>
|
||||||
</profile>
|
</profile>
|
||||||
|
|
||||||
|
|
|
||||||
|
|
@ -9,7 +9,7 @@
|
||||||
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
|
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
|
||||||
<!--<param name="from-domain" value="asterlink.com"/>-->
|
<!--<param name="from-domain" value="asterlink.com"/>-->
|
||||||
<!--/// account password *required* ///-->
|
<!--/// account password *required* ///-->
|
||||||
<!--<param name="password" value="2007"/>-->
|
<!--<param name="password" value="2007"/>-->
|
||||||
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
|
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
|
||||||
<!--<param name="extension" value="cluecon"/>-->
|
<!--<param name="extension" value="cluecon"/>-->
|
||||||
<!--/// proxy host: *optional* same as realm, if blank ///-->
|
<!--/// proxy host: *optional* same as realm, if blank ///-->
|
||||||
|
|
|
||||||
|
|
@ -1,14 +1,14 @@
|
||||||
<profile name="external">
|
<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 -->
|
<!-- This profile is only for outbound registrations to providers -->
|
||||||
<gateways>
|
<gateways>
|
||||||
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
|
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
|
||||||
</gateways>
|
</gateways>
|
||||||
|
|
||||||
<aliases>
|
<aliases>
|
||||||
<!--
|
<!--
|
||||||
<alias name="outbound"/>
|
<alias name="outbound"/>
|
||||||
<alias name="nat"/>
|
<alias name="nat"/>
|
||||||
-->
|
-->
|
||||||
</aliases>
|
</aliases>
|
||||||
|
|
||||||
|
|
@ -18,10 +18,10 @@
|
||||||
|
|
||||||
<settings>
|
<settings>
|
||||||
<param name="debug" value="0"/>
|
<param name="debug" value="0"/>
|
||||||
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
|
<!-- 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="shutdown-on-fail" value="true"/> -->
|
||||||
<param name="sip-trace" value="no"/>
|
<param name="sip-trace" value="no"/>
|
||||||
<param name="sip-capture" value="no"/>
|
<param name="sip-capture" value="no"/>
|
||||||
<param name="rfc2833-pt" value="101"/>
|
<param name="rfc2833-pt" value="101"/>
|
||||||
<!-- RFC 5626 : Send reg-id and sip.instance -->
|
<!-- RFC 5626 : Send reg-id and sip.instance -->
|
||||||
<!--<param name="enable-rfc-5626" value="true"/> -->
|
<!--<param name="enable-rfc-5626" value="true"/> -->
|
||||||
|
|
@ -34,15 +34,15 @@
|
||||||
<param name="hold-music" value="$${hold_music}"/>
|
<param name="hold-music" value="$${hold_music}"/>
|
||||||
<param name="rtp-timer-name" value="soft"/>
|
<param name="rtp-timer-name" value="soft"/>
|
||||||
<!--<param name="enable-100rel" value="true"/>-->
|
<!--<param name="enable-100rel" value="true"/>-->
|
||||||
<!--<param name="disable-srv503" value="true"/>-->
|
<!--<param name="disable-srv503" value="true"/>-->
|
||||||
<!-- This could be set to "passive" -->
|
<!-- This could be set to "passive" -->
|
||||||
<param name="local-network-acl" value="localnet.auto"/>
|
<param name="local-network-acl" value="localnet.auto"/>
|
||||||
<param name="manage-presence" value="false"/>
|
<param name="manage-presence" value="false"/>
|
||||||
|
|
||||||
<!-- used to share presence info across sofia profiles
|
<!-- used to share presence info across sofia profiles
|
||||||
manage-presence needs to be set to passive on this profile
|
manage-presence needs to be set to passive on this profile
|
||||||
if you want it to behave as if it were the internal profile
|
if you want it to behave as if it were the internal profile
|
||||||
for presence.
|
for presence.
|
||||||
-->
|
-->
|
||||||
<!-- Name of the db to use for this profile -->
|
<!-- Name of the db to use for this profile -->
|
||||||
<!--<param name="dbname" value="share_presence"/>-->
|
<!--<param name="dbname" value="share_presence"/>-->
|
||||||
|
|
@ -57,7 +57,7 @@
|
||||||
<param name="nonce-ttl" value="60"/>
|
<param name="nonce-ttl" value="60"/>
|
||||||
<param name="auth-calls" value="false"/>
|
<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="$${local_ip_v4}"/>
|
<param name="rtp-ip" value="$${local_ip_v4}"/>
|
||||||
<param name="sip-ip" value="$${local_ip_v4}"/>
|
<param name="sip-ip" value="$${local_ip_v4}"/>
|
||||||
|
|
@ -90,6 +90,5 @@
|
||||||
<param name="tls-verify-in-subjects" value=""/>
|
<param name="tls-verify-in-subjects" value=""/>
|
||||||
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
||||||
<param name="tls-version" value="$${sip_tls_version}"/>
|
<param name="tls-version" value="$${sip_tls_version}"/>
|
||||||
|
|
||||||
</settings>
|
</settings>
|
||||||
</profile>
|
</profile>
|
||||||
|
|
|
||||||
|
|
@ -9,7 +9,7 @@
|
||||||
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
|
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
|
||||||
<!--<param name="from-domain" value="asterlink.com"/>-->
|
<!--<param name="from-domain" value="asterlink.com"/>-->
|
||||||
<!--/// account password *required* ///-->
|
<!--/// account password *required* ///-->
|
||||||
<!--<param name="password" value="2007"/>-->
|
<!--<param name="password" value="2007"/>-->
|
||||||
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
|
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
|
||||||
<!--<param name="extension" value="cluecon"/>-->
|
<!--<param name="extension" value="cluecon"/>-->
|
||||||
<!--/// proxy host: *optional* same as realm, if blank ///-->
|
<!--/// proxy host: *optional* same as realm, if blank ///-->
|
||||||
|
|
|
||||||
|
|
@ -54,36 +54,36 @@
|
||||||
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
|
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
|
||||||
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
||||||
<param name="tls-version" value="$${sip_tls_version}"/>
|
<param name="tls-version" value="$${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)-->
|
<!--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="rtp-rewrite-timestamps" value="true"/>-->
|
||||||
<!--<param name="pass-rfc2833" value="true"/>-->
|
<!--<param name="pass-rfc2833" value="true"/>-->
|
||||||
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
|
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
|
||||||
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
|
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
|
||||||
|
|
||||||
<!--Uncomment to set all inbound calls to no media mode-->
|
<!--Uncomment to set all inbound calls to no media mode-->
|
||||||
<!--<param name="inbound-bypass-media" value="true"/>-->
|
<!--<param name="inbound-bypass-media" value="true"/>-->
|
||||||
|
|
||||||
<!--Uncomment to set all inbound calls to proxy media mode-->
|
<!--Uncomment to set all inbound calls to proxy media mode-->
|
||||||
<!--<param name="inbound-proxy-media" value="true"/>-->
|
<!--<param name="inbound-proxy-media" value="true"/>-->
|
||||||
|
|
||||||
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
|
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
|
||||||
<!--<param name="inbound-late-negotiation" value="true"/>-->
|
<!--<param name="inbound-late-negotiation" value="true"/>-->
|
||||||
|
|
||||||
<!-- this lets anything register -->
|
<!-- this lets anything register -->
|
||||||
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
|
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
|
||||||
<!-- <param name="accept-blind-reg" value="true"/> -->
|
<!-- <param name="accept-blind-reg" value="true"/> -->
|
||||||
|
|
||||||
<!-- accept any authentication without actually checking (not a good feature for most people) -->
|
<!-- accept any authentication without actually checking (not a good feature for most people) -->
|
||||||
<!-- <param name="accept-blind-auth" value="true"/> -->
|
<!-- <param name="accept-blind-auth" value="true"/> -->
|
||||||
|
|
||||||
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
|
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
|
||||||
<!-- <param name="suppress-cng" value="true"/> -->
|
<!-- <param name="suppress-cng" value="true"/> -->
|
||||||
|
|
||||||
<!--TTL for nonce in sip auth-->
|
<!--TTL for nonce in sip auth-->
|
||||||
<param name="nonce-ttl" value="60"/>
|
<param name="nonce-ttl" value="60"/>
|
||||||
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
|
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
|
||||||
that the originator is using-->
|
that the originator is using-->
|
||||||
<!--<param name="disable-transcoding" value="true"/>-->
|
<!--<param name="disable-transcoding" value="true"/>-->
|
||||||
<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
|
<!-- 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="NDLB-broken-auth-hash" value="true"/>-->
|
||||||
|
|
@ -103,8 +103,8 @@
|
||||||
<!-- <param name="vad" value="both"/> -->
|
<!-- <param name="vad" value="both"/> -->
|
||||||
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
|
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
|
||||||
<!--
|
<!--
|
||||||
These are enabled to make the default config work better out of the box.
|
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.
|
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 -->
|
<!--all inbound reg will look in this domain for the users -->
|
||||||
|
|
@ -121,10 +121,9 @@
|
||||||
<!-- set to true to have the profile determine stun is not useful and turn it off globally-->
|
<!-- set to true to have the profile determine stun is not useful and turn it off globally-->
|
||||||
<!--<param name="stun-auto-disable" value="true"/>-->
|
<!--<param name="stun-auto-disable" value="true"/>-->
|
||||||
|
|
||||||
<!-- the following can be used as workaround with bogus SRV/NAPTR records -->
|
<!-- the following can be used as workaround with bogus SRV/NAPTR records -->
|
||||||
<!--<param name="disable-srv" value="false" />-->
|
<!--<param name="disable-srv" value="false" />-->
|
||||||
<!--<param name="disable-naptr" value="false" />-->
|
<!--<param name="disable-naptr" value="false" />-->
|
||||||
|
|
||||||
</settings>
|
</settings>
|
||||||
</profile>
|
</profile>
|
||||||
|
|
||||||
|
|
|
||||||
|
|
@ -2,30 +2,30 @@
|
||||||
<!--
|
<!--
|
||||||
This is a sofia sip profile/user agent. This will service exactly one ip and port.
|
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.
|
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.
|
When you hear someone say "sofia profile" this is what they are talking about.
|
||||||
-->
|
-->
|
||||||
|
|
||||||
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
|
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
|
||||||
<!--aliases are other names that will work as a valid profile name for this profile-->
|
<!--aliases are other names that will work as a valid profile name for this profile-->
|
||||||
<aliases>
|
<aliases>
|
||||||
<!--
|
<!--
|
||||||
<alias name="default"/>
|
<alias name="default"/>
|
||||||
-->
|
-->
|
||||||
</aliases>
|
</aliases>
|
||||||
<!-- Outbound Registrations -->
|
<!-- Outbound Registrations -->
|
||||||
<gateways>
|
<gateways>
|
||||||
<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
|
<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
|
||||||
</gateways>
|
</gateways>
|
||||||
|
|
||||||
<domains>
|
<domains>
|
||||||
<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
|
<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
|
||||||
<!--<domain name="$${domain}" parse="true"/>-->
|
<!--<domain name="$${domain}" parse="true"/>-->
|
||||||
<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
|
<!-- 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="true"/>-->
|
||||||
<domain name="all" alias="true" parse="false"/>
|
<domain name="all" alias="true" parse="false"/>
|
||||||
</domains>
|
</domains>
|
||||||
|
|
||||||
<settings>
|
<settings>
|
||||||
|
|
||||||
|
|
||||||
|
|
@ -33,31 +33,31 @@
|
||||||
<!-- <param name="rtp-digit-delay" value="40"/>-->
|
<!-- <param name="rtp-digit-delay" value="40"/>-->
|
||||||
|
|
||||||
<!--
|
<!--
|
||||||
When calls are in no media this will bring them back to media
|
When calls are in no media this will bring them back to media
|
||||||
when you press the hold button.
|
when you press the hold button.
|
||||||
-->
|
-->
|
||||||
<!--<param name="media-option" value="resume-media-on-hold"/> -->
|
<!--<param name="media-option" value="resume-media-on-hold"/> -->
|
||||||
<!--
|
<!--
|
||||||
This will allow a call after an attended transfer go back to
|
This will allow a call after an attended transfer go back to
|
||||||
bypass media after an attended transfer.
|
bypass media after an attended transfer.
|
||||||
-->
|
-->
|
||||||
<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
|
<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
|
||||||
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
|
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
|
||||||
<param name="debug" value="0"/>
|
<param name="debug" value="0"/>
|
||||||
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
|
<!-- 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="shutdown-on-fail" value="true"/> -->
|
||||||
<param name="sip-trace" value="no"/>
|
<param name="sip-trace" value="no"/>
|
||||||
<param name="sip-capture" value="no"/>
|
<param name="sip-capture" value="no"/>
|
||||||
|
|
||||||
<!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
|
<!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
|
||||||
<!-- <param name="presence-proto-lookup" value="true"/> -->
|
<!-- <param name="presence-proto-lookup" value="true"/> -->
|
||||||
|
|
||||||
|
|
||||||
<!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
|
<!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
|
||||||
<!--<param name="liberal-dtmf" value="true"/>-->
|
<!--<param name="liberal-dtmf" value="true"/>-->
|
||||||
|
|
||||||
|
|
||||||
<!--
|
<!--
|
||||||
Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
|
Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
|
||||||
responding. These options allow you to enable and control a watchdog
|
responding. These options allow you to enable and control a watchdog
|
||||||
on the Sofia SIP stack so that if it stops responding for the
|
on the Sofia SIP stack so that if it stops responding for the
|
||||||
|
|
@ -70,7 +70,7 @@
|
||||||
through the FreeSWITCH CLI either on an individual profile basis or
|
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
|
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
|
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
|
If such crash occurs, FreeSWITCH will dump core if allowed. The
|
||||||
stacktrace will include function watchdog_triggered_abort().
|
stacktrace will include function watchdog_triggered_abort().
|
||||||
-->
|
-->
|
||||||
|
|
@ -106,26 +106,26 @@
|
||||||
|
|
||||||
<!--<param name="aggressive-nat-detection" value="true"/>-->
|
<!--<param name="aggressive-nat-detection" value="true"/>-->
|
||||||
<!--
|
<!--
|
||||||
There are known issues (asserts and segfaults) when 100rel is enabled.
|
There are known issues (asserts and segfaults) when 100rel is enabled.
|
||||||
It is not recommended to enable 100rel at this time.
|
It is not recommended to enable 100rel at this time.
|
||||||
-->
|
-->
|
||||||
<!--<param name="enable-100rel" value="true"/>-->
|
<!--<param name="enable-100rel" value="true"/>-->
|
||||||
|
|
||||||
<!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
|
<!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
|
||||||
<!-- RFC3263 Section 4.3 -->
|
<!-- RFC3263 Section 4.3 -->
|
||||||
<!--<param name="disable-srv503" value="true"/>-->
|
<!--<param name="disable-srv503" value="true"/>-->
|
||||||
|
|
||||||
<!-- Enable Compact SIP headers. -->
|
<!-- Enable Compact SIP headers. -->
|
||||||
<!--<param name="enable-compact-headers" value="true"/>-->
|
<!--<param name="enable-compact-headers" value="true"/>-->
|
||||||
<!--
|
<!--
|
||||||
enable/disable session timers
|
enable/disable session timers
|
||||||
-->
|
-->
|
||||||
<!--<param name="enable-timer" value="false"/>-->
|
<!--<param name="enable-timer" value="false"/>-->
|
||||||
<!--<param name="minimum-session-expires" value="120"/>-->
|
<!--<param name="minimum-session-expires" value="120"/>-->
|
||||||
<param name="apply-inbound-acl" value="domains"/>
|
<param name="apply-inbound-acl" value="domains"/>
|
||||||
<!--
|
<!--
|
||||||
This defines your local network, by default we detect your local network
|
This defines your local network, by default we detect your local network
|
||||||
and create this localnet.auto ACL for this.
|
and create this localnet.auto ACL for this.
|
||||||
-->
|
-->
|
||||||
<param name="local-network-acl" value="localnet.auto"/>
|
<param name="local-network-acl" value="localnet.auto"/>
|
||||||
<!--<param name="apply-register-acl" value="domains"/>-->
|
<!--<param name="apply-register-acl" value="domains"/>-->
|
||||||
|
|
@ -134,10 +134,10 @@
|
||||||
|
|
||||||
<!-- 'true' means every time 'first-only' means on the first register -->
|
<!-- 'true' means every time 'first-only' means on the first register -->
|
||||||
<!--<param name="send-message-query-on-register" value="true"/>-->
|
<!--<param name="send-message-query-on-register" value="true"/>-->
|
||||||
|
|
||||||
<!-- 'true' means every time 'first-only' means on the first register -->
|
<!-- 'true' means every time 'first-only' means on the first register -->
|
||||||
<!--<param name="send-presence-on-register" value="first-only"/> -->
|
<!--<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 -->
|
<!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
|
||||||
<!-- Remote-Party-ID header -->
|
<!-- Remote-Party-ID header -->
|
||||||
|
|
@ -164,7 +164,7 @@
|
||||||
<param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
|
<param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
|
||||||
<param name="presence-privacy" value="$${presence_privacy}"/>
|
<param name="presence-privacy" value="$${presence_privacy}"/>
|
||||||
<!-- ************************************************* -->
|
<!-- ************************************************* -->
|
||||||
|
|
||||||
<!-- This setting is for AAL2 bitpacking on G726 -->
|
<!-- This setting is for AAL2 bitpacking on G726 -->
|
||||||
<!-- <param name="bitpacking" value="aal2"/> -->
|
<!-- <param name="bitpacking" value="aal2"/> -->
|
||||||
<!--max number of open dialogs in proceeding -->
|
<!--max number of open dialogs in proceeding -->
|
||||||
|
|
@ -203,39 +203,39 @@
|
||||||
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
||||||
<param name="tls-version" value="$${sip_tls_version}"/>
|
<param name="tls-version" value="$${sip_tls_version}"/>
|
||||||
|
|
||||||
<!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
|
<!-- 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)-->
|
(reduces delay on latent connections default true, must be disabled explicitly)-->
|
||||||
<!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
|
<!--<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)-->
|
<!--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="rtp-rewrite-timestamps" value="true"/>-->
|
||||||
<!--<param name="pass-rfc2833" value="true"/>-->
|
<!--<param name="pass-rfc2833" value="true"/>-->
|
||||||
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
|
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
|
||||||
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
|
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
|
||||||
|
|
||||||
<!--Uncomment to set all inbound calls to no media mode-->
|
<!--Uncomment to set all inbound calls to no media mode-->
|
||||||
<!--<param name="inbound-bypass-media" value="true"/>-->
|
<!--<param name="inbound-bypass-media" value="true"/>-->
|
||||||
|
|
||||||
<!--Uncomment to set all inbound calls to proxy media mode-->
|
<!--Uncomment to set all inbound calls to proxy media mode-->
|
||||||
<!--<param name="inbound-proxy-media" value="true"/>-->
|
<!--<param name="inbound-proxy-media" value="true"/>-->
|
||||||
|
|
||||||
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
|
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
|
||||||
<!--<param name="inbound-late-negotiation" value="true"/>-->
|
<!--<param name="inbound-late-negotiation" value="true"/>-->
|
||||||
|
|
||||||
<!-- this lets anything register -->
|
<!-- this lets anything register -->
|
||||||
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
|
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
|
||||||
<!-- <param name="accept-blind-reg" value="true"/> -->
|
<!-- <param name="accept-blind-reg" value="true"/> -->
|
||||||
|
|
||||||
<!-- accept any authentication without actually checking (not a good feature for most people) -->
|
<!-- accept any authentication without actually checking (not a good feature for most people) -->
|
||||||
<!-- <param name="accept-blind-auth" value="true"/> -->
|
<!-- <param name="accept-blind-auth" value="true"/> -->
|
||||||
|
|
||||||
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
|
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
|
||||||
<!-- <param name="suppress-cng" value="true"/> -->
|
<!-- <param name="suppress-cng" value="true"/> -->
|
||||||
|
|
||||||
<!--TTL for nonce in sip auth-->
|
<!--TTL for nonce in sip auth-->
|
||||||
<param name="nonce-ttl" value="60"/>
|
<param name="nonce-ttl" value="60"/>
|
||||||
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
|
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
|
||||||
that the originator is using-->
|
that the originator is using-->
|
||||||
<!--<param name="disable-transcoding" value="true"/>-->
|
<!--<param name="disable-transcoding" value="true"/>-->
|
||||||
<!-- Handle 302 Redirect in the dialplan -->
|
<!-- Handle 302 Redirect in the dialplan -->
|
||||||
<!--<param name="manual-redirect" value="true"/> -->
|
<!--<param name="manual-redirect" value="true"/> -->
|
||||||
|
|
@ -252,16 +252,16 @@
|
||||||
<param name="inbound-reg-force-matching-username" value="true"/>
|
<param name="inbound-reg-force-matching-username" value="true"/>
|
||||||
<!-- on authed calls, authenticate *all* the packets not just invite -->
|
<!-- on authed calls, authenticate *all* the packets not just invite -->
|
||||||
<param name="auth-all-packets" value="false"/>
|
<param name="auth-all-packets" value="false"/>
|
||||||
|
|
||||||
<!-- external_sip_ip
|
<!-- external_sip_ip
|
||||||
Used as the public IP address for SDP.
|
Used as the public IP address for SDP.
|
||||||
Can be an one of:
|
Can be an one of:
|
||||||
ip address - "12.34.56.78"
|
ip address - "12.34.56.78"
|
||||||
a stun server lookup - "stun:stun.server.com"
|
a stun server lookup - "stun:stun.server.com"
|
||||||
a DNS name - "host:host.server.com"
|
a DNS name - "host:host.server.com"
|
||||||
auto - Use guessed ip.
|
auto - Use guessed ip.
|
||||||
auto-nat - Use ip learned from NAT-PMP or UPNP
|
auto-nat - Use ip learned from NAT-PMP or UPNP
|
||||||
-->
|
-->
|
||||||
<param name="ext-rtp-ip" value="auto-nat"/>
|
<param name="ext-rtp-ip" value="auto-nat"/>
|
||||||
<param name="ext-sip-ip" value="auto-nat"/>
|
<param name="ext-sip-ip" value="auto-nat"/>
|
||||||
|
|
||||||
|
|
@ -274,8 +274,8 @@
|
||||||
<!-- <param name="vad" value="both"/> -->
|
<!-- <param name="vad" value="both"/> -->
|
||||||
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
|
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
|
||||||
<!--
|
<!--
|
||||||
These are enabled to make the default config work better out of the box.
|
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.
|
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 -->
|
<!--all inbound reg will look in this domain for the users -->
|
||||||
|
|
@ -297,28 +297,27 @@
|
||||||
<!--<param name="disable-transfer" value="true"/>-->
|
<!--<param name="disable-transfer" value="true"/>-->
|
||||||
<!--<param name="disable-register" value="true"/>-->
|
<!--<param name="disable-register" value="true"/>-->
|
||||||
|
|
||||||
<!--
|
<!--
|
||||||
enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
|
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
|
right away, proxy waits until the call has been answered then sends accepts
|
||||||
-->
|
-->
|
||||||
<!--<param name="enable-3pcc" value="true"/>-->
|
<!--<param name="enable-3pcc" value="true"/>-->
|
||||||
|
|
||||||
<!-- use at your own risk or if you know what this does.-->
|
<!-- use at your own risk or if you know what this does.-->
|
||||||
<!--<param name="NDLB-force-rport" value="true"/>-->
|
<!--<param name="NDLB-force-rport" value="true"/>-->
|
||||||
<!--
|
<!--
|
||||||
Choose the realm challenge key. Default is auto_to if not set.
|
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.
|
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.
|
||||||
If you use any other value besides auto_to or auto_from you'll loose
|
<anyvalue> - you can input any value to use for the sip realm.
|
||||||
the ability to do multiple domains.
|
|
||||||
|
|
||||||
Note: comment out to restore the behavior before 2008-09-29
|
|
||||||
|
|
||||||
|
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="challenge-realm" value="auto_from"/>
|
||||||
<!--<param name="disable-rtp-auto-adjust" value="true"/>-->
|
<!--<param name="disable-rtp-auto-adjust" value="true"/>-->
|
||||||
|
|
@ -332,59 +331,58 @@
|
||||||
<!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
|
<!-- 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"/>-->
|
<!--<param name="pass-callee-id" value="false"/>-->
|
||||||
|
|
||||||
<!-- clear clears them all or supply the name to add or the name prefixed with ~ to remove
|
<!-- clear clears them all or supply the name to add or the name
|
||||||
valid values:
|
prefixed with ~ to remove valid values:
|
||||||
|
|
||||||
clear
|
clear
|
||||||
CISCO_SKIP_MARK_BIT_2833
|
CISCO_SKIP_MARK_BIT_2833
|
||||||
SONUS_SEND_INVALID_TIMESTAMP_2833
|
SONUS_SEND_INVALID_TIMESTAMP_2833
|
||||||
|
|
||||||
-->
|
-->
|
||||||
<!--<param name="auto-rtp-bugs" data="clear"/>-->
|
<!--<param name="auto-rtp-bugs" data="clear"/>-->
|
||||||
|
|
||||||
<!-- the following can be used as workaround with bogus SRV/NAPTR records -->
|
<!-- the following can be used as workaround with bogus SRV/NAPTR records -->
|
||||||
<!--<param name="disable-srv" value="false" />-->
|
<!--<param name="disable-srv" value="false" />-->
|
||||||
<!--<param name="disable-naptr" value="false" />-->
|
<!--<param name="disable-naptr" value="false" />-->
|
||||||
|
|
||||||
<!-- The following can be used to fine-tune timers within sofia's transport layer
|
<!-- 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 -->
|
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 -->
|
<!-- Initial retransmission interval (in milliseconds).
|
||||||
<!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
|
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" /> -->
|
||||||
|
|
||||||
|
|
||||||
<!-- By default mod_sofia will ignore the codecs in the sdp for hold/unhold operations
|
<!-- Maximum retransmission interval (in milliseconds).
|
||||||
Set this to true if you want to actually parse the sdp and re-negotiate the codec during hold/unhold.
|
Set the maximum retransmission interval used by the SIP transaction engine.
|
||||||
It's probably not what you want so stick with the default unless you really need to change this.
|
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
|
||||||
<!--<param name="renegotiate-codec-on-hold" value="true"/>-->
|
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>
|
</settings>
|
||||||
</profile>
|
</profile>
|
||||||
|
|
||||||
|
|
|
||||||
|
|
@ -9,7 +9,7 @@
|
||||||
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
|
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
|
||||||
<!--<param name="from-domain" value="asterlink.com"/>-->
|
<!--<param name="from-domain" value="asterlink.com"/>-->
|
||||||
<!--/// account password *required* ///-->
|
<!--/// account password *required* ///-->
|
||||||
<!--<param name="password" value="2007"/>-->
|
<!--<param name="password" value="2007"/>-->
|
||||||
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
|
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
|
||||||
<!--<param name="extension" value="cluecon"/>-->
|
<!--<param name="extension" value="cluecon"/>-->
|
||||||
<!--/// proxy host: *optional* same as realm, if blank ///-->
|
<!--/// proxy host: *optional* same as realm, if blank ///-->
|
||||||
|
|
|
||||||
Loading…
Reference in New Issue