net/freeswitch: Separated configs. Made -upstream-defaults config a separate package and added it to freeswitch-default. Also added -config-minimal and added it to freeswitch-minimal
git-svn-id: svn://svn.openwrt.org/openwrt/packages@21966 3c298f89-4303-0410-b956-a3cf2f4a3e73
This commit is contained in:
cshore
78
net/freeswitch/files/etc.minimal/sip_profiles/external.xml
Normal file
78
net/freeswitch/files/etc.minimal/sip_profiles/external.xml
Normal file
@ -0,0 +1,78 @@
|
||||
<profile name="external">
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<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
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||||
<!-- This profile is only for outbound registrations to providers -->
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<gateways>
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||||
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
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</gateways>
|
||||
|
||||
<aliases>
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||||
<!--
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<alias name="outbound"/>
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<alias name="nat"/>
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||||
-->
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||||
</aliases>
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||||
|
||||
<domains>
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<domain name="all" alias="false" parse="true"/>
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</domains>
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|
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<settings>
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<param name="debug" value="0"/>
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<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
|
||||
<!-- <param name="shutdown-on-fail" value="true"/> -->
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<param name="sip-trace" value="no"/>
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<param name="rfc2833-pt" value="101"/>
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<param name="sip-port" value="$${external_sip_port}"/>
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<param name="dialplan" value="XML"/>
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<param name="context" value="public"/>
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<param name="dtmf-duration" value="2000"/>
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<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
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<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
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<param name="hold-music" value="$${hold_music}"/>
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<param name="rtp-timer-name" value="soft"/>
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||||
<!--<param name="enable-100rel" value="true"/>-->
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||||
<!-- This could be set to "passive" -->
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<param name="local-network-acl" value="localnet.auto"/>
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<param name="manage-presence" value="false"/>
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||||
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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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if you want it to behave as if it were the internal profile
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for presence.
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-->
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<!-- Name of the db to use for this profile -->
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<!--<param name="dbname" value="share_presence"/>-->
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<!--<param name="presence-hosts" 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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<!--<param name="force-register-db-domain" value="$${domain}"/>-->
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<!-- ************************************************* -->
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|
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<!--<param name="aggressive-nat-detection" value="true"/>-->
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<param name="inbound-codec-negotiation" value="generous"/>
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<param name="nonce-ttl" value="60"/>
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<param name="auth-calls" value="false"/>
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<!--
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DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
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-->
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<param name="rtp-ip" value="$${local_ip_v4}"/>
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<param name="sip-ip" value="$${local_ip_v4}"/>
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<param name="ext-rtp-ip" value="auto-nat"/>
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<param name="ext-sip-ip" value="auto-nat"/>
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<param name="rtp-timeout-sec" value="300"/>
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<param name="rtp-hold-timeout-sec" value="1800"/>
|
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<!--<param name="enable-3pcc" value="true"/>-->
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|
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<!-- TLS: disabled by default, set to "true" to enable -->
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<param name="tls" value="$${external_ssl_enable}"/>
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<!-- additional bind parameters for TLS -->
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<param name="tls-bind-params" value="transport=tls"/>
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<!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
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<param name="tls-sip-port" value="$${external_tls_port}"/>
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<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
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<param name="tls-cert-dir" value="$${external_ssl_dir}"/>
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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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</settings>
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</profile>
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38
net/freeswitch/files/etc.minimal/sip_profiles/external/example.xml
vendored
Normal file
38
net/freeswitch/files/etc.minimal/sip_profiles/external/example.xml
vendored
Normal file
@ -0,0 +1,38 @@
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<include>
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<!-- Don't change gateway name unless you also change the dialplan -->
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<gateway name="example.com">
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<!--/// account username *required* ///-->
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<!--<param name="username" value="cluecon"/>-->
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<!--/// auth realm: *optional* same as gateway name, if blank ///-->
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<!--<param name="realm" value="asterlink.com"/>-->
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<!--/// username to use in from: *optional* same as username, if blank ///-->
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<!--<param name="from-user" value="cluecon"/>-->
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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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<!--/// account password *required* ///-->
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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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<!--<param name="extension" value="cluecon"/>-->
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<!--/// proxy host: *optional* same as realm, if blank ///-->
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<!--<param name="proxy" value="asterlink.com"/>-->
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<!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
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<!--<param name="register-proxy" value="mysbc.com"/>-->
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<!--/// expire in seconds: *optional* 3600, if blank ///-->
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<!--<param name="expire-seconds" value="60"/>-->
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<!--/// should this profile regiser with the gateway? ///-->
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<!--<param name="register" value="false"/>-->
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<!-- which transport to use for register -->
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<!--<param name="register-transport" value="udp"/>-->
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<!--How many seconds before a retry when a failure or timeout occurs -->
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<!--<param name="retry-seconds" value="30"/>-->
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<!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
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<!--<param name="caller-id-in-from" value="false"/>-->
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<!--extra sip params to send in the contact-->
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<!--<param name="contact-params" value="tport=tcp"/>-->
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<!--/// Force the contact paramater to a particular value (e.g. extension number). ///-->
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<!--/// Use this if your gateways are registering at gw+username@ip and this is a problem for your provider. ///-->
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<!--<param name="extension-in-contact" value="Xxxx" />-->
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<!--send an options ping every x seconds, failure will unregister and/or mark it down-->
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<!--<param name="ping" value="25"/>-->
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</gateway>
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</include>
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296
net/freeswitch/files/etc.minimal/sip_profiles/internal.xml
Normal file
296
net/freeswitch/files/etc.minimal/sip_profiles/internal.xml
Normal file
@ -0,0 +1,296 @@
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||||
<profile name="internal">
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||||
<!--
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||||
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.
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||||
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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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||||
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
|
||||
<!--aliases are other names that will work as a valid profile name for this profile-->
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<aliases>
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<!--
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<alias name="default"/>
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-->
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||||
</aliases>
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||||
<!-- Outbound Registrations -->
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||||
<gateways>
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||||
<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
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||||
</gateways>
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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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<!--<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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<!--<domain name="all" alias="true" parse="true"/>-->
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<domain name="all" alias="true" parse="false"/>
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</domains>
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||||
<settings>
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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 you press the hold button.
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||||
-->
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||||
<!--<param name="media-option" value="resume-media-on-hold"/> -->
|
||||
<!--
|
||||
This will allow a call after an attended transfer go back to
|
||||
bypass media after an attended transfer.
|
||||
-->
|
||||
<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
|
||||
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
|
||||
<param name="debug" value="0"/>
|
||||
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
|
||||
<!-- <param name="shutdown-on-fail" value="true"/> -->
|
||||
<param name="sip-trace" value="no"/>
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||||
<param name="log-auth-failures" value="true"/>
|
||||
<param name="context" value="public"/>
|
||||
<param name="rfc2833-pt" value="101"/>
|
||||
<!-- port to bind to for sip traffic -->
|
||||
<param name="sip-port" value="$${internal_sip_port}"/>
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<param name="dialplan" value="XML"/>
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||||
<param name="dtmf-duration" value="2000"/>
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||||
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
|
||||
<param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
|
||||
<param name="rtp-timer-name" value="soft"/>
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<!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
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<param name="rtp-ip" value="$${local_ip_v4}"/>
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<!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
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||||
<param name="sip-ip" value="$${local_ip_v4}"/>
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||||
<param name="hold-music" value="$${hold_music}"/>
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||||
<param name="apply-nat-acl" value="nat.auto"/>
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||||
|
||||
<!-- extended info parsing -->
|
||||
<!-- <param name="extended-info-parsing" value="true"/> -->
|
||||
|
||||
<!--<param name="aggressive-nat-detection" value="true"/>-->
|
||||
<!--
|
||||
There are known issues (asserts and segfaults) when 100rel is enabled.
|
||||
It is not recommended to enable 100rel at this time.
|
||||
-->
|
||||
<!--<param name="enable-100rel" value="true"/>-->
|
||||
<!-- Enable Compact SIP headers. -->
|
||||
<!--<param name="enable-compact-headers" value="true"/>-->
|
||||
<!--
|
||||
enable/disable session timers
|
||||
-->
|
||||
<!--<param name="enable-timer" value="false"/>-->
|
||||
<!--<param name="minimum-session-expires" value="120"/>-->
|
||||
<param name="apply-inbound-acl" value="domains"/>
|
||||
<!--
|
||||
This defines your local network, by default we detect your local network
|
||||
and create this localnet.auto ACL for this.
|
||||
-->
|
||||
<param name="local-network-acl" value="localnet.auto"/>
|
||||
<!--<param name="apply-register-acl" value="domains"/>-->
|
||||
<!--<param name="dtmf-type" value="info"/>-->
|
||||
|
||||
|
||||
<!-- 'true' means every time 'first-only' means on the first register -->
|
||||
<!--<param name="send-message-query-on-register" value="true"/>-->
|
||||
|
||||
|
||||
|
||||
<param name="record-path" value="$${recordings_dir}"/>
|
||||
<param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
|
||||
<!--enable to use presence -->
|
||||
<param name="manage-presence" value="true"/>
|
||||
<!--<param name="manage-shared-appearance" value="true"/>-->
|
||||
<!-- used to share presence info across sofia profiles -->
|
||||
<!-- Name of the db to use for this profile -->
|
||||
<!--<param name="dbname" value="share_presence"/>-->
|
||||
<!--<param name="presence-hosts" value="$${domain}"/>-->
|
||||
<!-- ************************************************* -->
|
||||
|
||||
<!-- This setting is for AAL2 bitpacking on G726 -->
|
||||
<!-- <param name="bitpacking" value="aal2"/> -->
|
||||
<!--max number of open dialogs in proceeding -->
|
||||
<!--<param name="max-proceeding" value="1000"/>-->
|
||||
<!--session timers for all call to expire after the specified seconds -->
|
||||
<!--<param name="session-timeout" value="120"/>-->
|
||||
<!-- Can be 'true' or 'contact' -->
|
||||
<!--<param name="multiple-registrations" value="contact"/>-->
|
||||
<!--set to 'greedy' if you want your codec list to take precedence -->
|
||||
<param name="inbound-codec-negotiation" value="generous"/>
|
||||
<!-- if you want to send any special bind params of your own -->
|
||||
<!--<param name="bind-params" value="transport=udp"/>-->
|
||||
<!--<param name="unregister-on-options-fail" value="true"/>-->
|
||||
|
||||
<!-- TLS: disabled by default, set to "true" to enable -->
|
||||
<param name="tls" value="$${internal_ssl_enable}"/>
|
||||
<!-- additional bind parameters for TLS -->
|
||||
<param name="tls-bind-params" value="transport=tls"/>
|
||||
<!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
|
||||
<param name="tls-sip-port" value="$${internal_tls_port}"/>
|
||||
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
|
||||
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
|
||||
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
|
||||
<param name="tls-version" value="$${sip_tls_version}"/>
|
||||
|
||||
<!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
|
||||
(reduces delay on latent connections default true, must be disabled explicitly)-->
|
||||
<!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
|
||||
|
||||
<!--If you don't want to pass through timestamps from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
|
||||
<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
|
||||
<!--<param name="pass-rfc2833" value="true"/>-->
|
||||
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
|
||||
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
|
||||
|
||||
<!--Uncomment to set all inbound calls to no media mode-->
|
||||
<!--<param name="inbound-bypass-media" value="true"/>-->
|
||||
|
||||
<!--Uncomment to set all inbound calls to proxy media mode-->
|
||||
<!--<param name="inbound-proxy-media" value="true"/>-->
|
||||
|
||||
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
|
||||
<!--<param name="inbound-late-negotiation" value="true"/>-->
|
||||
|
||||
<!-- this lets anything register -->
|
||||
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
|
||||
<!-- <param name="accept-blind-reg" value="true"/> -->
|
||||
|
||||
<!-- accept any authentication without actually checking (not a good feature for most people) -->
|
||||
<!-- <param name="accept-blind-auth" value="true"/> -->
|
||||
|
||||
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
|
||||
<!-- <param name="suppress-cng" value="true"/> -->
|
||||
|
||||
<!--TTL for nonce in sip auth-->
|
||||
<param name="nonce-ttl" value="60"/>
|
||||
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
|
||||
that the originator is using-->
|
||||
<!--<param name="disable-transcoding" value="true"/>-->
|
||||
<!-- Handle 302 Redirect in the dialplan -->
|
||||
<!--<param name="manual-redirect" value="true"/> -->
|
||||
<!-- Disable Transfer -->
|
||||
<!--<param name="disable-transfer" value="true"/> -->
|
||||
<!-- Disable Register -->
|
||||
<!--<param name="disable-register" value="true"/> -->
|
||||
<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
|
||||
<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
|
||||
<!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
|
||||
<!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
|
||||
<param name="auth-calls" value="$${internal_auth_calls}"/>
|
||||
<!-- Force the user and auth-user to match. -->
|
||||
<param name="inbound-reg-force-matching-username" value="true"/>
|
||||
<!-- on authed calls, authenticate *all* the packets not just invite -->
|
||||
<param name="auth-all-packets" value="false"/>
|
||||
|
||||
<!-- external_sip_ip
|
||||
Used as the public IP address for SDP.
|
||||
Can be an one of:
|
||||
ip address - "12.34.56.78"
|
||||
a stun server lookup - "stun:stun.server.com"
|
||||
a DNS name - "host:host.server.com"
|
||||
auto - Use guessed ip.
|
||||
auto-nat - Use ip learned from NAT-PMP or UPNP
|
||||
-->
|
||||
<param name="ext-rtp-ip" value="auto-nat"/>
|
||||
<param name="ext-sip-ip" value="auto-nat"/>
|
||||
|
||||
<!-- rtp inactivity timeout -->
|
||||
<param name="rtp-timeout-sec" value="300"/>
|
||||
<param name="rtp-hold-timeout-sec" value="1800"/>
|
||||
<!-- VAD choose one (out is a good choice); -->
|
||||
<!-- <param name="vad" value="in"/> -->
|
||||
<!-- <param name="vad" value="out"/> -->
|
||||
<!-- <param name="vad" value="both"/> -->
|
||||
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
|
||||
<!--
|
||||
These are enabled to make the default config work better out of the box.
|
||||
If you need more than ONE domain you'll need to not use these options.
|
||||
|
||||
-->
|
||||
<!--all inbound reg will look in this domain for the users -->
|
||||
<param name="force-register-domain" value="$${domain}"/>
|
||||
<!--force the domain in subscriptions to this value -->
|
||||
<param name="force-subscription-domain" value="$${domain}"/>
|
||||
<!--all inbound reg will stored in the db using this domain -->
|
||||
<param name="force-register-db-domain" value="$${domain}"/>
|
||||
<!--force suscription expires to a lower value than requested-->
|
||||
<!--<param name="force-subscription-expires" value="60"/>-->
|
||||
<!-- disable register and transfer which may be undesirable in a public switch -->
|
||||
<!--<param name="disable-transfer" value="true"/>-->
|
||||
<!--<param name="disable-register" value="true"/>-->
|
||||
|
||||
<!--
|
||||
enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
|
||||
right away, proxy waits until the call has been answered then sends accepts
|
||||
-->
|
||||
<!--<param name="enable-3pcc" value="true"/>-->
|
||||
|
||||
<!-- use at your own risk or if you know what this does.-->
|
||||
<!--<param name="NDLB-force-rport" value="true"/>-->
|
||||
<!--
|
||||
Choose the realm challenge key. Default is auto_to if not set.
|
||||
|
||||
auto_from - uses the from field as the value for the sip realm.
|
||||
auto_to - uses the to field as the value for the sip realm.
|
||||
<anyvalue> - you can input any value to use for the sip realm.
|
||||
|
||||
If you want URL dialing to work you'll want to set this to auto_from.
|
||||
|
||||
If you use any other value besides auto_to or auto_from you'll loose
|
||||
the ability to do multiple domains.
|
||||
|
||||
Note: comment out to restore the behavior before 2008-09-29
|
||||
|
||||
-->
|
||||
<param name="challenge-realm" value="auto_from"/>
|
||||
<!--<param name="disable-rtp-auto-adjust" value="true"/>-->
|
||||
<!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
|
||||
<!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
|
||||
<!-- on outbound calls set the callid to match the uuid of the session -->
|
||||
<!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
|
||||
<!-- set to false disable this feature -->
|
||||
<!--<param name="rtp-autofix-timing" value="false"/>-->
|
||||
|
||||
<!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
|
||||
<!--<param name="pass-callee-id" value="false"/>-->
|
||||
|
||||
<!-- clear clears them all or supply the name to add or the name prefixed with ~ to remove
|
||||
valid values:
|
||||
|
||||
clear
|
||||
CISCO_SKIP_MARK_BIT_2833
|
||||
SONUS_SEND_INVALID_TIMESTAMP_2833
|
||||
|
||||
-->
|
||||
<!--<param name="auto-rtp-bugs" data="clear"/>-->
|
||||
|
||||
<!-- the following can be used as workaround with bogus SRV/NAPTR records -->
|
||||
<!--<param name="disable-srv" value="false" />-->
|
||||
<!--<param name="disable-naptr" value="false" />-->
|
||||
|
||||
<!-- The following can be used to fine-tune timers within sofia's transport layer
|
||||
Those settings are for advanced users and can safely be left as-is -->
|
||||
|
||||
<!-- Initial retransmission interval (in milliseconds).
|
||||
Set the T1 retransmission interval used by the SIP transaction engine.
|
||||
The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. -->
|
||||
<!-- <param name="timer-T1" value="500" /> -->
|
||||
|
||||
<!-- Transaction timeout (defaults to T1 * 64).
|
||||
Set the T1x64 timeout value used by the SIP transaction engine.
|
||||
The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine.
|
||||
The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
|
||||
<!-- <param name="timer-T1X64" value="32000" /> -->
|
||||
|
||||
|
||||
<!-- Maximum retransmission interval (in milliseconds).
|
||||
Set the maximum retransmission interval used by the SIP transaction engine.
|
||||
The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine.
|
||||
Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially
|
||||
until the timer B fires. -->
|
||||
<!-- <param name="timer-T2" value="4000" /> -->
|
||||
|
||||
<!--
|
||||
Transaction lifetime (in milliseconds).
|
||||
Set the lifetime for completed transactions used by the SIP transaction engine.
|
||||
A completed transaction is kept around for the duration of T4 in order to catch late responses.
|
||||
The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
|
||||
<!-- <param name="timer-T4" value="4000" /> -->
|
||||
|
||||
</settings>
|
||||
</profile>
|
||||
|
||||
@ -0,0 +1,36 @@
|
||||
<include>
|
||||
<!--<gateway name="asterlink.com">-->
|
||||
<!--/// account username *required* ///-->
|
||||
<!--<param name="username" value="cluecon"/>-->
|
||||
<!--/// auth realm: *optional* same as gateway name, if blank ///-->
|
||||
<!--<param name="realm" value="asterlink.com"/>-->
|
||||
<!--/// username to use in from: *optional* same as username, if blank ///-->
|
||||
<!--<param name="from-user" value="cluecon"/>-->
|
||||
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
|
||||
<!--<param name="from-domain" value="asterlink.com"/>-->
|
||||
<!--/// account password *required* ///-->
|
||||
<!--<param name="password" value="2007"/>-->
|
||||
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
|
||||
<!--<param name="extension" value="cluecon"/>-->
|
||||
<!--/// proxy host: *optional* same as realm, if blank ///-->
|
||||
<!--<param name="proxy" value="asterlink.com"/>-->
|
||||
<!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
|
||||
<!--<param name="register-proxy" value="mysbc.com"/>-->
|
||||
<!--/// expire in seconds: *optional* 3600, if blank ///-->
|
||||
<!--<param name="expire-seconds" value="60"/>-->
|
||||
<!--/// do not register ///-->
|
||||
<!--<param name="register" value="false"/>-->
|
||||
<!-- which transport to use for register -->
|
||||
<!--<param name="register-transport" value="udp"/>-->
|
||||
<!--How many seconds before a retry when a failure or timeout occurs -->
|
||||
<!--<param name="retry-seconds" value="30"/>-->
|
||||
<!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
|
||||
<!--<param name="caller-id-in-from" value="false"/>-->
|
||||
<!--extra sip params to send in the contact-->
|
||||
<!--<param name="contact-params" value="tport=tcp"/>-->
|
||||
<!-- Put the extension in the contact -->
|
||||
<!--<param name="extension-in-contact" value="true"/>-->
|
||||
<!--send an options ping every x seconds, failure will unregister and/or mark it down-->
|
||||
<!--<param name="ping" value="25"/>-->
|
||||
<!--</gateway>-->
|
||||
</include>
|
||||
Reference in New Issue
Block a user