<div dir="ltr">Hi Steven, thanks for your reply.<div><br></div><div>Yes, you are right, I had it all backwards :( </div><div><br></div><div>I solved this by bridging to a gateway but <b><u>also</u></b> had to place this before bridging the call:</div><div><div><br></div><div> <action application="set" data="bypass_media=true"/></div></div><div><br></div><div>Thanks again for replying.</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Nov 22, 2016 at 6:14 PM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Are you sure 'internal' is the sofia profile you want to be using to reach an external address?</div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="h5">On 21 November 2016 at 13:35, Cato Gonzalez <span dir="ltr"><<a href="mailto:catogonzalez@gmail.com" target="_blank">catogonzalez@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div dir="ltr">Hi everyone,<div><br></div><div>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:</div><div><br></div><div><div> <action application="bridge" data="sofia/internal/<a href="mailto:XXX@sip.linphone.org" target="_blank">XXX@sip.l<wbr>inphone.org</a>"/></div></div><div><br></div><div>I have a softphone registered to this provider <a href="http://sip.linphone.org" target="_blank">sip.linphone.org</a> and it works well when calling XXX from some other IP phone. FS reports this in the logs:</div><div><br></div><div><div>[DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@48000hz 1 channel 20ms</div><div>[DEBUG] switch_core_codec.c:221 sofia/internal/<a href="mailto:anonymous@webrtc.sip.mydomain.co" target="_blank">anonymous@webrt<wbr>c.sip.mydomain.co</a> Push codec L16:100</div><div>[DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)]</div><div>[DEBUG] sofia.c:6760 Channel sofia/internal/<a href="mailto:anonymous@webrtc.sip.mydomain.co" target="_blank">anonymous@webrt<wbr>c.sip.mydomain.co</a> entering state [terminated][500]</div><div>[NOTICE] sofia.c:7779 Hangup sofia/internal/<a href="mailto:anonymous@webrtc.sip.mydomain.co" target="_blank">anonymous@webrt<wbr>c.sip.mydomain.co</a> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]</div><div>[DEBUG] switch_core_codec.c:246 sofia/internal/<a href="mailto:anonymous@webrtc.sip.mydomain.co" target="_blank">anonymous@webrt<wbr>c.sip.mydomain.co</a> Restore previous codec opus:116.</div><div>[NOTICE] switch_ivr_originate.c:3523 Hangup sofia/external/<a href="mailto:XXX@sip.linphone.org" target="_blank">XXX@sip.linphon<wbr>e.org</a> [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]</div></div><div><br></div><div>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.</div><div><br></div><div>Thanks,</div><div><br></div><div>Cato</div><div><br></div><div><br></div><div><br></div></div>
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