Could you collect the full siptrace of one call and paste into pastebin?<div><br></div><div>To do so, type at the cli:</div><div><br></div><div> sofia global siptrace on</div><div><br></div><div>Then make 1 call and collect whatever you got at the screen, and paste into pastebin.</div>
<div><br></div><div>Then, you'll probably want to disable siptrace:</div><div><br></div><div> sofia global siptrace off</div><div><br></div><div>Regards,</div><div>Vitalie</div><div><br></div><div><br><div class="gmail_quote">
2012/4/12 Malay Thakershi <span dir="ltr"><<a href="mailto:mthakershi@gmail.com">mthakershi@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Thanks.<br>
<br>
Before I do what you said, I have another problem. Sorry about that.<br>
<br>
I use different mechanism to make outgoing calls (originate calls from<br>
ESL). So I do not use bridge.<br>
I tried to put "{absolute_codec_string=GSM}" in originate call along<br>
with other parameters I pass but it throws error.<br>
<br>
My originate call looks like:<br>
originate {absolute_codec_string=GSM,ignore_early_media=true,origination_caller_id_name=CH,origination_caller_id_number=1877XXXXXXX,originate_timeout=45,chivr_outbound=true,other_var=1}sofia/gateway/vitelity-outbound/1972XXXXXXX<br>
1056 XML default<br>
<br>
Error thrown on the log: -ERR INCOMPATIBLE_DESTINATION<br>
2012-04-12 19:14:01.589739 [NOTICE] sofia.c:6332 Hangup<br>
sofia/sipinterface_1/1972XXXXXXX [CS_CONSUME_MEDIA]<br>
[INCOMPATIBLE_DESTINATION]<br>
<br>
Call without absolute_codec_string works fine.<br>
<br>
The way this works is: call gets originated -- it goes to default dial<br>
plan extension 1056 -- mod_managed is called from this point to handle<br>
call flow.<br>
<br>
<extension name="CH1"><br>
<!-- Calls from public.xml will come with 1056 extension and will<br>
be handled here --><br>
<condition field="destination_number" expression="^1056$"><br>
<!-- codec trial setting --><br>
<!-- <action application="set" data="absolute_codec_string=GSM"/> --><br>
<!-- <action application="log" data="codec debug: codec_string is<br>
${codec_string}" /><br>
<action application="log" data="codec debug: originator_codec is<br>
${originator_codec}" /><br>
<action application="log" data="codec debug: absolute_codec_string<br>
is ${absolute_codec_string}" /><br>
--><br>
<div class="im"> <action application="sleep" data="1000" /><br>
<action application="managed" data="clsAsmtApp" /><br>
</condition><br>
</extension><br>
<br>
</div>As you can see, I tried setting absolute string from here too but it<br>
didn't work.<br>
<br>
Let me know what can be done if you can.<br>
<br>
Thanks.<br>
<span class="HOEnZb"><font color="#888888"><br>
Malay<br>
</font></span><div class="HOEnZb"><div class="h5"><br>
<br>
On Wed, Apr 11, 2012 at 5:48 PM, Vitalie Colosov <<a href="mailto:vetali100@gmail.com">vetali100@gmail.com</a>> wrote:<br>
> Lets forget for a minute about vars.xml and what is there (when you use<br>
> absolute_codec_string=GSM, it overrides any settings from vars.xml), and<br>
> concentrate at the OUTGOING call from FreeSWITCH to your sip provider first.<br>
><br>
> Try to change your bridge to the following string:<br>
> <action application="bridge"><br>
> data="{absolute_codec_string=GSM}sofia/gateway/your_provider/12345678901_example"/><br>
><br>
> (obviously use your sip provider and phone number :)<br>
><br>
> And do one outgoing call to this sip provider.<br>
><br>
> Then, reply back whether the call was successful or not (did you hear the<br>
> voice and whether the other party did hear you). And paste the log to the<br>
> pastebin (<a href="http://pastebin.freeswitch.org/" target="_blank">http://pastebin.freeswitch.org/</a>) and reply back with the link.<br>
><br>
><br>
><br>
><br>
><br>
> 2012/4/11 Malay Thakershi <<a href="mailto:mthakershi@gmail.com">mthakershi@gmail.com</a>><br>
>><br>
>> > 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to<br>
>> > use<br>
>> > G729 right now as it needs the license). By OUTGOING I mean call from<br>
>> > FreeSWITCH to your provider, and not back.<br>
>> Yes. I want to try some codec that will use less bandwidth and see how<br>
>> much audio quality is compromised. So I chose GSM because G729 has<br>
>> license requirements.<br>
>><br>
>> I want this to work when someone calls my DID or my FS program calls<br>
>> someone via DID. So both for incoming calls as well as outgoing calls.<br>
>><br>
>> > 2. If #1 is true, then inbound-codec-negotiation is not much useful at<br>
>> > this point. Because this setting is for INCOMING leg, and not for<br>
>> > OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but<br>
>> > still<br>
>> > transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need<br>
>> > to<br>
>> > find out - why).<br>
>> I am not very familiar with internals of FS so having lot of trouble<br>
>> figuring this out.<br>
>><br>
>> Weird thing is, even if I do not have any mention of G729, it is being<br>
>> negotiated and used. Here are my configuration lines:<br>
>> vars.xml<br>
>> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/><br>
>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/><br>
>><br>
>> sip profile has:<br>
>> <param name="inbound-codec-negotiation" value="greedy"/><br>
>><br>
>> dial plan<br>
>> <extension name="CH"><br>
>> <condition field="destination_number" expression="^105\d$"><br>
>> <action application="set"<br>
>> data="absolute_codec_string=GSM"/><br>
>> <action application="sleep" data="1000"/><br>
>> <action application="managed" data="clsAsmtApp"/><br>
>> </condition><br>
>> </extension><br>
>><br>
>><br>
>> <extension name="vitel-outbound"><br>
>> <condition field="destination_number" expression="^(1{0,1}\d{10})$"><br>
>> <action application="set" data="absolute_codec_string=GSM"/><br>
>> <action application="set"<br>
>> data="effective_caller_id_number=18770001111"/><br>
>> <action application="bridge"<br>
>> data="sofia/gateway/vitelity-outbound/$1"/><br>
>> </condition><br>
>> </extension><br>
>><br>
>> > 3. Try using absolute_codec_string before bridging your call to your SIP<br>
>> > provider:<br>
>> ><br>
>> > <action application="set" data="absolute_codec_string=GSM"/><br>
>> > <action application="bridge"<br>
>> > data="sofia/gateway/your_provider/12345678901_example"/><br>
>> ><br>
>> I did as you said. But found what I described above. If I remove<br>
>> "greedy" setting, system simply uses PCMU codec like before.<br>
>><br>
>> This thing is so confusing. Even my SIP provider is saying they can't<br>
>> help out with FS because some guy quit who knew the configurations.<br>
>><br>
>> Thanks for help.<br>
>><br>
>> On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <<a href="mailto:vetali100@gmail.com">vetali100@gmail.com</a>><br>
>> wrote:<br>
>> ><br>
>> > Lets confirm the following statements now:<br>
>> ><br>
>> > 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to<br>
>> > use<br>
>> > G729 right now as it needs the license). By OUTGOING I mean call from<br>
>> > FreeSWITCH to your provider, and not back.<br>
>> ><br>
>> > 2. If #1 is true, then inbound-codec-negotiation is not much useful at<br>
>> > this point. Because this setting is for INCOMING leg, and not for<br>
>> > OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but<br>
>> > still<br>
>> > transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need<br>
>> > to<br>
>> > find out - why).<br>
>> ><br>
>> > 3. Try using absolute_codec_string before bridging your call to your SIP<br>
>> > provider:<br>
>> ><br>
>> > <action application="set" data="absolute_codec_string=GSM"/><br>
>> > <action application="bridge"<br>
>> > data="sofia/gateway/your_provider/12345678901_example"/><br>
>> ><br>
>> ><br>
>> > And reply back if this changes anything...<br>
>> ><br>
>> > Vitalie<br>
>> ><br>
>> > 2012/4/11 Malay Thakershi <<a href="mailto:mthakershi@gmail.com">mthakershi@gmail.com</a>><br>
>> >><br>
>> >> Under my sipinterface_1 profile that I use for these calls, I have:<br>
>> >><br>
>> >> <param name="inbound-codec-negotiation" value="greedy"/><br>
>> >><br>
>> >> On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <<a href="mailto:lazyvirus@gmx.com">lazyvirus@gmx.com</a>> wrote:<br>
>> >>><br>
>> >>> On Wed, 11 Apr 2012 13:31:45 -0500<br>
>> >>> Malay Thakershi <<a href="mailto:mthakershi@gmail.com">mthakershi@gmail.com</a>> wrote:<br>
>> >>><br>
>> >>> > Understood your point about G722. But if my list has GSM at higher<br>
>> >>> > priority<br>
>> >>> > over PCMU/PCMA and far-end SDP supports GSM then why isn't the call<br>
>> >>> > using<br>
>> >>> > GSM?<br>
>> >>> ><br>
>> >>> > I don't know where G726 comparisons are coming from. If I haven't<br>
>> >>> > mentioned<br>
>> >>> > it in vars.xml and far-end SDP doesn't support it then why would it<br>
>> >>> > even<br>
>> >>> > consider that codec?<br>
>> >>><br>
>> >>> From: <a href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files" target="_blank">wiki.freeswitch.org/wiki/Sofia_Configuration_Files</a><br>
>> >>> (inbound-codec-negotiation):<br>
>> >>> 'generous' permits the remote codec list have precedence and 'win'<br>
>> >>> the codec negotiation and selection process<br>
>> >>><br>
>> >>> JY<br>
>> >>> --<br>
>> >>> Don't drop acid -- take it pass/fail.<br>
>> >>> -- Seen in a Ladies' Room at Harvard<br>
>> >>><br>
>> >>><br>
>> >>><br>
>> >>> _________________________________________________________________________<br>
>> >>> Professional FreeSWITCH Consulting Services:<br>
>> >>> <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
>> >>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
>> >>><br>
>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
>> >>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
>> >>><br>
>> >>> Official FreeSWITCH Sites<br>
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>> >>><br>
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>> >>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>> >>><br>
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>> >>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >><br>
>> >><br>
>> >><br>
>> >><br>
>> >> _________________________________________________________________________<br>
>> >> Professional FreeSWITCH Consulting Services:<br>
>> >> <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
>> >> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
>> >><br>
>> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
>> >> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
>> >><br>
>> >> Official FreeSWITCH Sites<br>
>> >> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
>> >> <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
>> >><br>
>> >> FreeSWITCH-users mailing list<br>
>> >> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>> >> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>> >><br>
>> >> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
>> >> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> >><br>
>> ><br>
>> ><br>
>> ><br>
>> > _________________________________________________________________________<br>
>> > Professional FreeSWITCH Consulting Services:<br>
>> > <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
>> > <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
>> ><br>
>> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
>> > <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
>> ><br>
>> > Official FreeSWITCH Sites<br>
>> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> > <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
>> > <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
>> ><br>
>> > FreeSWITCH-users mailing list<br>
>> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>> > UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
>> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> ><br>
>><br>
>> _________________________________________________________________________<br>
>> Professional FreeSWITCH Consulting Services:<br>
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>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
>><br>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
>><br>
>> Official FreeSWITCH Sites<br>
>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
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>><br>
>> FreeSWITCH-users mailing list<br>
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>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
><br>
><br>
><br>
> _________________________________________________________________________<br>
> Professional FreeSWITCH Consulting Services:<br>
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> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
><br>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
><br>
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><br>
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><br>
<br>
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</div></div></blockquote></div><br></div>