[Freeswitch-users] absolute_codec_string question

[Nicolas Brenner]

Thanks, yes, here you go:

I'm making calls from the console to an X-Lite registered on extension 1001
which doesn't support G729. Calls 1 and 3 below fail because the codec
options are not supported, and even though on the third call the
absolute_codec_string variable should be 'G729,PCMU', it is not offering
both codecs.


On http://pastebin.freeswitch.org/16167 the SIP trace for:

originate {ignore_early_media,absolute_codec_string=G729}user/1001
&bridge(user/1000)

This is the SDP:

   v=0
   o=FreeSWITCH 1303734153 1303734154 IN IP4 127.0.0.1
   s=FreeSWITCH
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 24036 RTP/AVP 18 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20


On http://pastebin.freeswitch.org/16169 the SIP trace for:

originate {ignore_early_media,absolute_codec_string=PCMU}user/1001
&bridge(user/1000)

This is the SDP:

   v=0
   o=FreeSWITCH 1303733515 1303733516 IN IP4 127.0.0.1
   s=FreeSWITCH
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 24896 RTP/AVP 0 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20


On http://pastebin.freeswitch.org/16170 the SIP trace for:

originate {ignore_early_media,absolute_codec_string='G729,PCMU'}user/1001
&bridge(user/1000)

This is the SDP:

   v=0
   o=FreeSWITCH 1303733520 1303733521 IN IP4 127.0.0.1
   s=FreeSWITCH
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 25138 RTP/AVP 18 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20




On Mon, Apr 25, 2011 at 2:20 PM, Paul Cupis <paul at cupis.co.uk> wrote:

> On 25/04/11 18:08, Nicolas Brenner wrote:
> > I'm trying to use the
> > absolute_codec_string with originate from the console, like so:
> >
> > originate
> >
> {ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444
> > &bridge(user/1001)
> >
> > Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the same
> as
> > if I don't quote the string, or if I just specify one codec:
>
> Can you provide (on pastebin.freeswitch.org) a complete log of a call,
> please?
>
> Regards,
>
>
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