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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