Yes, the rate of the telephone-event must be the rate of the audio codec. Out of curiosity, what gateway is this ?
As much as I hate the idea, you must not offer Opus to the B-leg in this case.

You can try this just before you bridge:
<action application="set" data="codec_string=OPUS,PCMA"/>
<action application="set" data="inbound_codec_prefs=OPUS,PCMA"/>
<action application="export" data="nolocal:absolute_codec_string=PCMA"/>

Cheers,
Dragos
 

On Wed, Nov 1, 2017 at 9:08 AM, Zheka Polivoda <poliv78@yahoo.co.uk> wrote:
Hi all
I'm using freeswitch to call from browser(webrtc) to some sip provider.
So, the A-keg is webrtc opus codec and then I bridge it with the sip gateway. 
Sip gateway supports g711 only. And it chooses wrong codec and DTMF is not working between them.
When bridging to sip, freeswitch sends such SDP


72a02659-6c1b-424e-b76a-0a9b16aecc2a Local SDP:
72a02659-6c1b-424e-b76a-0a9b16aecc2a v=0
72a02659-6c1b-424e-b76a-0a9b16aecc2a o=FreeSWITCH 1509346805 1509346806 IN IP4 x.x.x.x
72a02659-6c1b-424e-b76a-0a9b16aecc2a s=FreeSWITCH
72a02659-6c1b-424e-b76a-0a9b16aecc2a c=IN IP4 x.x.x.x
72a02659-6c1b-424e-b76a-0a9b16aecc2a t=0 0
72a02659-6c1b-424e-b76a-0a9b16aecc2a m=audio 31568 RTP/AVP 102 9 0 8 103 101
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:102 opus/48000/2
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; sprop-maxcapturerate=8000; ptime=20; minptime=10; maxptim
e=40; stereo=1
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:9 G722/8000
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:0 PCMU/8000
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:8 PCMA/8000
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:103 telephone-event/48000
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:103 0-16
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:101 telephone-event/8000
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:101 0-16
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=ptime:20
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=sendrecv



Is it possible to send SDP without opus rtpmap? Because SIP gateway choose wrong 
telephone-event. the answer of sip gateway is 

72a02659-6c1b-424e-b76a-0a9b16aecc2a 2017-10-30 15:46:13.248384 [DEBUG] sofia.c:7058 Remote SDP:
72a02659-6c1b-424e-b76a-0a9b16aecc2a v=0
72a02659-6c1b-424e-b76a-0a9b16aecc2a o=lvp 8000 8000 IN IP4 y.y.y.y
72a02659-6c1b-424e-b76a-0a9b16aecc2a s=SIP Call
72a02659-6c1b-424e-b76a-0a9b16aecc2a c=IN IP4 y.y.y.y
72a02659-6c1b-424e-b76a-0a9b16aecc2a t=0 0
72a02659-6c1b-424e-b76a-0a9b16aecc2a m=audio 22786 RTP/AVP 8 103
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:8 PCMA/8000
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:103 telephone-event/48000
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:103 0-16
72a02659-6c1b-424e-b76a-0a9b16aecc2a a=ptime:20


I guess there must be 
a=rtpmap:101 telephone-event/8000

Thanks

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