I installed asterisk 16 and I’m having some problems with receiving PJSIP calls. I think it would be easier to gradually migrate from chan_sip.so to res_pjsip.
IS THERE A POSSIBILITY OF EXISTING SIP AND PJSIP ON THE SAME ASTERISK?
hmm yes it is possible
but for it to work you need to have in depth knowledge about both chan_sip and chan_pjsip
it is fare easier to get just one of the to work, getting both to work is like chan_sip squared chan_pjsip
Yes, For example FreePBX does it. You need to bind to different signalling ports, of course.
However, if the problem is the one in your other thread, your problem isn’t with the channel driver but rather with a rather strange configuration of the peer, which make dialplans difficult to write.
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