Lets forget for a minute about <span style>vars.xml</span> and what is there (when you use
absolute_codec_string=GSM, it overrides any settings from vars.xml), and concentrate at the OUTGOING call from FreeSWITCH to your sip provider first.<div><br></div><div>Try to change your bridge to the following string:</div>
<div><action application="bridge"> data="{absolute_codec_string=GSM}sofia/gateway/your_provider/12345678901_example"/></div><div><br></div><div>(obviously use your sip provider and phone number :) </div>
<div><br></div><div>And do one outgoing call to this sip provider.</div><div><br></div><div>Then, reply back whether the call was successful or not (did you hear the voice and whether the other party did hear you). And paste the log to the pastebin (<a href="http://pastebin.freeswitch.org/">http://pastebin.freeswitch.org/</a>) and reply back with the link.</div>
<div><br></div><div><br></div><div><br></div><div><br></div><div><br><div class="gmail_quote">2012/4/11 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">
<div class="im">> 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to 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>
</div>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>
<div class="im"><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 still<br>
> transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need to<br>
> find out - why).<br>
</div>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>
<div class="im"><param name="inbound-codec-negotiation" value="greedy"/><br>
<br>
</div>dial plan<br>
<extension name="CH"><br>
<condition field="destination_number" expression="^105\d$"><br>
<div class="im"> <action application="set" data="absolute_codec_string=GSM"/><br>
</div> <action application="sleep" data="1000"/><br>
<action application="managed" data="clsAsmtApp"/><br>
</condition><br>
</extension><br>
<br>
<br>
<extension name="vitel-outbound"><br>
<condition field="destination_number" expression="^(1{0,1}\d{10})$"><br>
<div class="im"> <action application="set" data="absolute_codec_string=GSM"/><br>
</div> <action application="set" data="effective_caller_id_number=18770001111"/><br>
<action application="bridge" data="sofia/gateway/vitelity-outbound/$1"/><br>
</condition><br>
</extension><br>
<div class="im"><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>
</div>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>
<div class="HOEnZb"><div class="h5"><br>
On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <<a href="mailto:vetali100@gmail.com">vetali100@gmail.com</a>> 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 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 still<br>
> transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need 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>
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</div></div></blockquote></div><br></div>