<div dir="ltr"><div>Well, it seems my mail thread management has failed me. I totally missed KK's post here. At least we gave him the same info. :P<br></div>-MC<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Fri, Jun 6, 2014 at 11:15 AM, Kristian Kielhofner <span dir="ltr"><<a href="mailto:kris@kriskinc.com" target="_blank">kris@kriskinc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
The codecs were included from the start (except GSM):<br>
<div class=""><br>
m=audio 17262 RTP/AVP 8 0 101 13<br>
<br>
</div>8 = PCMA<br>
0 = PCMU<br>
101 = telephone-event (dynamic)<br>
13 = CNG<br>
<br>
See this:<br>
<br>
<a href="https://wiki.freeswitch.org/wiki/Variable_verbose_sdp" target="_blank">https://wiki.freeswitch.org/wiki/Variable_verbose_sdp</a><br>
<br>
You might not be seeing GSM because it's not included on the profile<br>
or the module (mod_spandsp, I believe) isn't loaded.<br>
<div class="HOEnZb"><div class="h5"><br>
<br>
On Fri, Jun 6, 2014 at 9:51 AM, Ričardas Stoma <<a href="mailto:ricardas.stoma@gmail.com">ricardas.stoma@gmail.com</a>> wrote:<br>
> It is still the same without single quote marks around PCMA,PCMU,GSM...<br>
><br>
> But i think everything is OK with my SDP, i just made a call to my asterisk<br>
> server and i see that asterisk receives correct codecs:<br>
><br>
> Found audio description format telephone-event for ID 101<br>
> Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer -<br>
> audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined -<br>
> (gsm|ulaw|alaw)<br>
><br>
> So i guess i had to learn more about SDP and codecs before posting my issue,<br>
> sorry guys :-)<br>
><br>
><br>
> 2014-06-06 15:47 GMT+03:00 Anonio Silva <<a href="mailto:asilva@wirelessmundi.com">asilva@wirelessmundi.com</a>>:<br>
><br>
>> Can you try:<br>
>><br>
>> {absolute_codec_string=PCMA,PCMU,GSM} ===> without '<br>
>><br>
>><br>
>><br>
>> On Fri, 2014-06-06 at 11:36 +0300, Ričardas Stoma wrote:<br>
>><br>
>> I have created the following extension:<br>
>><br>
>><br>
>> <extension name="my_test"><br>
>> <condition><br>
>> <action application="bridge"<br>
>> data="{absolute_codec_string='PCMA,PCMU,GSM'}sofia/external/<a href="mailto:1001@192.168.1.78">1001@192.168.1.78</a>"/><br>
>> <action application="hangup"/><br>
>> </condition><br>
>> </extension><br>
>><br>
>><br>
>> then i call to this extension, freeswitch dials to <a href="mailto:1001@192.168.1.78">1001@192.168.1.78</a> and<br>
>> when i check the content of SDP, included in a SIP packet, i see something<br>
>> like this:<br>
>><br>
>><br>
>> v=0<br>
>> o=FreeSWITCH 1401972565 1401972566 IN IP4 192.168.1.65<br>
>> s=FreeSWITCH<br>
>> c=IN IP4 192.168.1.65<br>
>> t=0 0<br>
>> m=audio 17262 RTP/AVP 8 0 101 13<br>
>> a=rtpmap:101 telephone-event/8000<br>
>> a=fmtp:101 0-16<br>
>> a=ptime:20<br>
>><br>
>><br>
>> why my codecs from 'absolute_codec_string' are not included in SDP? I<br>
>> would expect to see these lines included:<br>
>><br>
>> a=rtpmap:8 PCMA/8000<br>
>> a=rtpmap:0 PCMU/8000<br>
>> a=rtpmap:3 GSM/8000<br>
>><br>
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>><br>
><br>
><br>
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><br>
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><br>
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><br>
<br>
<br>
<br>
</div></div><span class="HOEnZb"><font color="#888888">--<br>
Kristian Kielhofner<br>
</font></span><div class="HOEnZb"><div class="h5"><br>
_________________________________________________________________________<br>
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<br>
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</div></div></blockquote></div><br></div>