I am having a small problem that is driving me nuts. I can make calls over my Webrtc client without any problems and audio sounds fine. The only problem I have is that when I call an internal SIP extension on my PBX I do not hear the ring while I wait for the call to be answered. My dial command does include the rR options. If I make an external call to a land line or a mobile phone I do hear the ring sounds, only internal extensions have this problem. Why would the webrtc client ignore the ringing…
I have a requirement to dialout some external number, once the call is answered the same has to be forwarded to an Internal Queue.
Please help me.
I have tried calling with two SIP end point forwarding , even that is not working,
My dial plan line is , Dial(SIP/19201/19202,300)
I am trying to set add a SIP Header to a call before adding it to the Queue.
The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs (verbose 999). In the SIP case, I see it.
Does the function Set(PJSIP_HEADER(add, ..... not transfer over to the call when the Queue…
I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset. I captured the SIP traffic and see that my M65 is replying with an "488 not acceptable here". From what I
read this is usually codec related but both asterisk and the M65 are set for ulaw as first choice.
I have a SIP trace below. Can someone suggest why the 488 is being generated?