[Freeswitch-users] Bridge an incoming call to an external SIP address

[Cato Gonzalez]

Hi everyone,

I am trying to get FS to route inbound calls to an outside sip provider but
I am getting the call hung up on the FS side just upon the first or second
ring on the B-leg. The diaplan action being executed is:

         <action application="bridge" data="sofia/internal/
XXX at sip.linphone.org"/>

I have a softphone registered to this provider sip.linphone.org and it
works well when calling XXX from some other IP phone. FS reports this in
the logs:

[DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16 at 48000hz
1 channel 20ms
[DEBUG] switch_core_codec.c:221 sofia/internal/
anonymous at webrtc.sip.mydomain.co Push codec L16:100
[DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone
[%(2000,4000,440,480)]
[DEBUG] sofia.c:6760 Channel sofia/internal/anonymous at webrtc.sip.mydomain.co
entering state [terminated][500]
[NOTICE] sofia.c:7779 Hangup sofia/internal/anonymous at webrtc.sip.mydomain.co
[CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
[DEBUG] switch_core_codec.c:246 sofia/internal/
anonymous at webrtc.sip.mydomain.co Restore previous codec opus:116.
[NOTICE] switch_ivr_originate.c:3523 Hangup sofia/external/
XXX at sip.linphone.org [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]

Result is the same if I try to call any other sip destination. I appreciate
any input anyone may give: have been working on this for a week already.

Thanks,

Cato
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