I am working on a problem where about 70% of incoming calls from telasip.com are being disconnected immediately after I answer.
I have an asterisk server with a idefisk softphone and a SIP trunk to telasip for incoming calls. The asterisk server is on a private address behind a cisco PIX doing PAT on the outside interface. I am using dynamic dns. I have set my rtp.conf file to use ports 10001 through 10015. My firewall has a static translation from the outside interface to the internal IP of the server for udp 10001 - 10015 and udp 5060.
Sometimes it works fine but again, about 70% of the time the call is disconnected immediately after I answer it.
any ideas?
well I’ve never restricted the range to such a narrow band, plus I believe rtp is only even ports, so you are effectively 10002-10014, I would suspect that. Is it always to the same phone? The symptom sounds similar to codec mismatches (do you see anyting in the log file that says something like ‘can’t make channels compatible’? Which is often a symptom? If it isn’t a codec mismatch, I would definitely broaden your rtp range to something like 10k-20k and see if you have the same problems. (telasip does a reinvite once the call is established so it sounds like it migh be failing at that point and could be related to the rtp port range.)
You could do a sip debug and look at the sip messages to see if the reinvite is trying to use ports that you don’t have available, or otherwise gives a clue as to who and why the call is being ended.
p