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

[Steven Ayre]

Are you sure 'internal' is the sofia profile you want to be using to reach
an external address?

On 21 November 2016 at 13:35, Cato Gonzalez <catogonzalez at gmail.com> wrote:

> 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@
> 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@
> webrtc.sip.mydomain.co entering state [terminated][500]
> [NOTICE] sofia.c:7779 Hangup sofia/internal/anonymous@
> webrtc.sip.mydomain.co [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> [DEBUG] switch_core_codec.c:246 sofia/internal/anonymous@
> 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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