Hi,

I am trying to make an outbound call to a webrtc softphone using jssip, I initiate the call from an asterisk box :

[Asterisk] -> [FS] ->[jssip]

I always get an  INCOMPATIBLE_DESTINATION error, looking at the trace logs I found out that the problem is the codec negotiation but I can not make it work, AFAIK the call should use the ALAW codec as it is compatible with all legs involved. this is the asterisk SDP

2013-08-06 13:16:50.892293 [DEBUG] sofia.c:5802 Remote SDP:
v=0
o=root 451068671 451068671 IN IP4 192.168.90.16
s=Asterisk PBX 1.8.16.0
c=IN IP4 192.168.90.16
t=0 0
m=audio 16126 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

And this is the SDP from FS wich is sent to the jssip client

2013-08-06 13:16:50.912323 [DEBUG] sofia_glue.c:1220 Local SDP:
v=0
o=FreeSWITCH 1375766038 1375766039 IN IP4 PUBLIC_IP
s=FreeSWITCH
c=IN IP4 PUBLIC_IP
t=0 0
a=msid-semantic: WMS 9hJmtW2hnx2uG5S2obryUBUKGF35tgps
m=audio 21772 RTP/SAVPF 8 0 101 13
a=rtpmap:101 telephone-event/8000
a=rtcp-mux
a=rtcp:21772 IN IP4 PUBLIC_IP
a=ssrc:3456358538 cname:oZFIMxPZUIkrD3jS
a=ssrc:3456358538 msid:9hJmtW2hnx2uG5S2obryUBUKGF35tgps a0
a=ssrc:3456358538 mslabel:9hJmtW2hnx2uG5S2obryUBUKGF35tgps
a=ssrc:3456358538 label:9hJmtW2hnx2uG5S2obryUBUKGF35tgpsa0
a=ice-ufrag:4BgP20qzvjuPqK7E
a=ice-pwd:ADcc0S2Lqc2T5cB0
a=candidate:3263618716 1 udp 659136 212.230.135.231 21772 typ host generation 0
a=candidate:3263618716 2 udp 659136 212.230.135.231 21772 typ host generation 0
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:JbAOcaClBcoNLW6zjoucy5BU0mfS+UQqWcyYh9+7
a=ptime:20
a=sendrecv

Shouldn't it include the PCMA codec? I hace tried to enable it in configuration but it doesn't work, also tried to set it with

  <action application="export" data="nolocal:absolute_codec_string=PCMA,PCMU"/>

before bridge, but no luck.

Funny thing is that it is working if i innitiate the call from the jssip client. [jssip] -> [FS] ->[Asterisk]

Any clue?

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