[Freeswitch-users] Freeswitch not processing calls

[Ankit Gandhi]

Hi Brian,
But issue here is that, FS is not processing any such calls and not sending
488 to the caller.
Also the sip trace I had provided was from the caller to fs. FS does not
even bridge the call to terminator in between the INVITE and CANCEL from the
caller.
It just gives so many errors in log like following :

2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is
only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is
only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is
only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is
only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is
only usable in passthrough mode!

It seems odd that due to codec mismatch, fs does not process the call at all
and just times out and the caller sends CANCEL to us. 
Waiting for your reply.


Brian West-3 wrote:
> 
> I notice it offers 18 which is G729 but these listed below are 100%  
> invalid.  There is no such thing as G.729a, G.729b or G.729ab that are  
> valid in the SDP.  I suspect if you start FreeSWITCH and crank it up  
> to debug level ("console loglevel debug") you'll clearly see why this  
> is taking place.  I think we talked to you on IRC about this and told  
> this.  If your termination gateway requires any of the above listed on  
> 96,97 or 98 the call will fail.  We can only do passthru on G729 which  
> is the official name which is listed on payload 18.  Your termination  
> provider needs to be informed of this fact.  The fmtp line is what  
> controls annexb operation!
> 
> /b
> 
> On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote:
> 
>> a=rtpmap:97 G.729b/8000.
>> a=rtpmap:98 G.729a/8000.
>> a=rtpmap:96 G.729ab/8000.
> 
> 
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