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I enabled tcp on PBX, added sip_force_tcp=true and now calls can be established from WCS to Bria. But it still goes on hold(Bria saying that call is on hold by other party). I added a new report if you could please take a look. Is there some asterisk/pjsip configuration to enforce a=sendrecv...

None of parameters worked. I can't give SSH access. You can please try to test on https://demo.flashphoner.com/ ? using my sip server/accounts or yours. I reported my sip details. Our end goal is to call from 'SIP as RTMP" page (first account) to "SIP Phone" page (second account) and listen but...

I can't find a solution. Only configuration I found that is related to sendrecv is moh_passthrough=yes and I added it to pjsip endpoints. I can make calls from flashphoner phone to flasphoner phone, but I still can't make calls from flashphoner (phone or REST call) to any softphones(microsip...

Hello, I am trying to integrate Asterisk pjsip with flashphoner. I am going to page SIP -> Phone I register successfully, but I don't receive calls in that page and if I call from that page to zoiper call goes on hold. I sent logs to support mail and also used report button.