~25 Polycom 430s
Each call whether coming in over the PRI or ext <> ext loses audio (on the far side (callee)) for about 5s. This loss of audio happens almost exclusively about 3-5s from the beginning of the call.
Once audio is re-established the call continues with no further issues.
The issue occurs whether coming from a queue, ring group, or ext <> ext.
The phones are all on a local network, gigabit POE switch, and a centOS linux router acting as the gateway. I have fiddled with the NAT settings thinking this may have been the issue to no avail.
From what I am reading it looks like chan_sip uses 5061 now? I have adjusted the phone configs to use 5061 as well as the extension’s Advanced --> Port config. I have fiddled with the NAT settings thinking this may have been the issue to no avail.
I am still seeing the following:
`<— SIP read from UDP:10.10.10.101:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.7:5061;branch=z9hG4bK667b80b5
From: “Unknown” sip:Unknown@10.10.10.7:5061;tag=as42c9cc2e
CSeq: 102 NOTIFY
My suspicion is that I am losing audio when the RTP stream is setup/switched over?
 deny=0.0.0.0/0.0.0.0 secret=XXXXXXXXXXXX dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=no port=5061 qualify=yes qualifyfreq=60 transport=udp,tcp,tls avpf=no force_avp=no icesupport=no encryption=no namedcallgroup= namedpickupgroup= dial=SIP/200 mailbox=200@default permit=0.0.0.0/0.0.0.0 callerid=Front Desk <200> callcounter=yes faxdetect=no
I can send/pb any logging output you need. I don’t see anything that sticks out in the debug log.
Any help would be appreciated.