[Freeswitch-users] Issue with a=fmtp:18 annexb=no

[Sergey Okhapkin]

Nevermind, my suggestion will work for bridging calls, but no for answering.

On Thursday 13 February 2014 10:43:45 Sergey Okhapkin wrote:
> application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"
> 
> On Thursday 13 February 2014 07:09:16 Sam wrote:
> > Hi,
> > 
> > We are facing an issue with a=fmtp:18 annexb=no.
> > 
> > Here is the call flow,
> > Cisco CUCM ->  Cisco Cube -> FS
> > 
> > Cisco is expecting in 200 OK ,
> > 
> >    m=audio 28820 RTP/AVP 18 101 13
> >    a=rtpmap:18 G729/8000
> >    * a=fmtp:18 annexb=no*
> >    a=rtpmap:101 telephone-event/8000
> >    a=fmtp:101 0-16
> >    a=ptime:20
> > 
> > But fs is generating without  *a=fmtp:18 annexb=no*
> > 
> >    m=audio 28820 RTP/AVP 18 101 13
> >    a=rtpmap:18 G729/8000
> >    a=rtpmap:101 telephone-event/8000
> >    a=fmtp:101 0-16
> >    a=ptime:20
> > 
> > Because of that calls are getting fail.
> > 
> > I am not getting where should we change setting to include that line.
> > 
> > Log,
> > http://pastebin.com/LDHEmCNv
> > 
> > Please help me to resolve that issue.
> > 
> > Thanks in advance.
> > 
> > 
> > 
> > -----
> > Thanks,
> > Sam
> > --
> > View this message in context:
> > http://freeswitch-users.2379917.n2.nabble.com/Issue-with-a-fmtp-18-annexb-> > n
> > o-tp7596060.html Sent from the freeswitch-users mailing list archive at
> > Nabble.com.
> > 
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