We are using the following external sip profile:

<profile name="external">

  <settings>

    <param name="accept-blind-auth" value="true"/>

    <param name="apply-inbound-acl" value="domains"/>

    <param name="debug" value="0"/>

    <param name="sip-trace" value="no"/>

    <param name="rfc2833-pt" value="101"/>

    <param name="sip-port" value="5080"/>

    <param name="dialplan" value="XML"/>

    <param name="context" value="public"/>

    <param name="dtmf-duration" value="2000"/>

    <param name="inbound-codec-prefs" value="G729,PCMU"/>

    <param name="outbound-codec-prefs" value="G729,PCMU"/>

    <param name="hold-music" value="$${hold_music}"/>

    <param name="rtp-timer-name" value="soft"/>

    <param name="manage-presence" value="false"/>

    <param name="inbound-codec-negotiation" value="generous"/>

    <param name="nonce-ttl" value="60"/>

    <param name="auth-calls" value="false"/>

    <param name="rtp-ip" value="10.10.3.12"/>

    <param name="sip-ip" value="10.10.3.12"/>

    <param name="rtp-timeout-sec" value="300"/>

    <param name="rtp-hold-timeout-sec" value="1800"/>

    <param name="user-agent-string" value="INTEGRAL PARTNERS"/>

  </settings>

</profile>

The dial plan “public” context has two entries, one for PSTN access and one for call routing to an internal PBX.

This is the dial plan for PSTN call routing:

<extension name="LCL">

   <condition field="destination_number" expression="^(\d{10})$">

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$1@172.16.15.11"/>

       <action application="set" data="ringback=${us-ring}"/>

       <action application="set" data="ignore_early_media=true" />

   </condition>

</extension>

<extension name="LD">

   <condition field="destination_number" expression="^1(\d{10})$">

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$0@172.16.15.11"/>

       <action application="set" data="ringback=${us-ring}"/>

       <action application="set" data="ignore_early_media=true" />

   </condition>

</extension>

<extension name="911">

   <condition field="destination_number" expression="^(911)$">

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$1@172.16.15.11"/>

       <action application="set" data="ringback=${us-ring}"/>

       <action application="set" data="ignore_early_media=true" />

   </condition>

</extension>

<extension name="INTL">

   <condition field="destination_number" expression="^011(\d+)$">

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$0@172.16.15.11"/>

       <action application="set" data="ringback=${us-ring}"/>

       <action application="set" data="ignore_early_media=true" />

   </condition>

</extension>

And this is the entry for internal call routing to a PBX:

<extension name="PBX">

   <condition field="destination_number" expression="^(205314849[0-9])$">

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$1@10.10.3.10"|data="sofia/external/$1@10.10.3.11"/>

       <action application="set" data="ringback=${us-ring}"/>

       <action application="set" data="ignore_early_media=true"/>

   </condition>

</extension>

So, here’s my question:

We’d like to be able to lock down the codec as 711 for the internal leg going to the PBX and 729 for the external leg to the PSTN. We have transcoding setup and it’s working fine. How can we use two SIP profiles  to hard code the codec in each direction? I’ve seen an example in the dial plan section, but didn’t understand how to implement it.

Thanks,

Ken Fulmer