[Freeswitch-users] Problem with Freeswitch/Sangoma D100.

[David Villasmil]

Hello Ricardo,

Did you figure out what your problem was?

Thanks

David

On Fri, Jul 22, 2011 at 5:56 PM, Ricardo Martinez <rmartinez at redvoiss.net>wrote:

> Hello list.
>
> I have the next problem with my Freeswitch.  I’m using a Sangoma D100
> transcoding card, but ‘im having problems with G729.  This is my scenario.
>
>
>
> GatewayA ---------> FreeSwitch+D100 -----------> GatewayB
>
> (10.0.0.220)                    (10.0.0.148)
>       (10.0.0.222)
>
>
>
> Gateway A is calling through Freeswitch to Gateway B.  The problem is with
> the “200 - OK” message that goes from FreeSwitch to Gateway A.  The “200 -
> OK”  coming from Gateway B has the parameter “a=fmtp:18 annexb=no” in the
> SDP, but Freeswitch is not attaching this parameter to the “Leg A” and this
> is causing a one-way audio in the call.  (please see the attached
> sdp-problem.jpg file).
>
> The weird thing is when i unload the “mod_sangoma_codec” and load the
> “mod_g729”, this time the “200 OK” message from the Freeswitch is using the
> “a=fmtp:18 annexb=no” to the Leg A, and the call is successfully established
> without one way audio problem.
>
> I really don’t know what could be happening, I ask the Sangoma support but
> thay said that this is a Freeswitch bug.
>
> Can someone help me here?
>
>
>
> These are part of my configuration files.
>
>
>
> “default.xml”
>
>
>
>     <extension name="to_prueba2">
>
>       <condition field="network_addr" expression="^10\.0\.0\.220$">
>
>                   <action application="set" data="call_timeout=50"/>
>
>                   <action application="set"
> data="hangup_after_bridge=true"/>
>
>                   <action application="export"
> data="nolocal:absolute_codec_string=G729,G723"/>
>
>                   <action application="set"
> data="sip_invite_domain=10.0.0.222"/>
>
>                   <action application="export"
> data="sip_append_audio_sdp=a=rtpmap:18 G729/8000,a=fmtp:18 annexb=no"/>
>
>                   <action application="bridge" data="
> sofia/$${domain}/${destination_number}@10.0.0.222"/>
>
>                   <action application="answer"/>
>
>       </condition>
>
>     </extension>
>
>
>
>     <extension name="to_prueba1">
>
>       <condition field="network_addr" expression="^10\.0\.0\.222$">
>
>                   <action application="set" data="call_timeout=50"/>
>
>                   <action application="set"
> data="hangup_after_bridge=true"/>
>
>                   <action application="set"
> data="sip_invite_domain=10.0.0.220"/>
>
>                   <action application="bridge" data="
> sofia/$${domain}/${destination_number}@10.0.0.220"/>
>
>                   <action application="answer"/>
>
>       </condition>
>
>     </extension>
>
>
>
>
>
> Part of the  “interior.xml” file:
>
>     <!--Uncomment to let calls hit the dialplan *before* you decide if the
> codec is ok-->
>
>     <param name="inbound-late-negotiation" value="true"/>
>
>     <!--set to 'greedy' if you want your codec list to take precedence -->
>
>     <param name="inbound-codec-negotiation" value="generous"/>
>
>
>
> Y finalmente en los codec preferentes tengo (vars.xml)
>
>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G723,PCMU,PCMA"/>
>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,G723"/>
>
>
>
>
>
>
>
> Thanks in advance.
>
> Regards,
>
> Ricardo.-
>
>
>
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