I am in the process of setting up some inter-site trunking for our company using Asterisk and IAX as the transport over the WAN. We have a Mitel 3300 one side, and a Nortel CS1000 on the other, e.g.:
[color=white]…[/color]Site 1[color=white]…[/color]WAN[color=white]…[/color]Site 2
[Mitel]<-sip->[Asterisk]<--------IAX-------->[Asterisk]<-sip->[Nortel]
Everything on Site 1 side works just fine. Asterisk to Asterisk works as it should. A user on the Mitel can call a user on the Nortel by their extension number and that is working ok.
The problem comes going the other way. If someone on the Nortel calls someone on the Mitel, the call lasts approx 30 seconds, then drops out. I have checked the link by making and receiving calls a sip device registered to the Site 2 Asterisk box - all works fine.
Debug output:
-- Executing [1513203@default:1] Dial("SIP/wapp.com-08253720", "IAX2/trunk_1/13203") in new stack
-- Called trunk_1/13203
-- Call accepted by xxx.xxx.xxx.xxx (format gsm)
-- Format for call is gsm
-- IAX2/trunk_1-4 is ringing
-- IAX2/trunk_1-4 stopped sounds
-- IAX2/trunk_1-4 answered SIP/wapp.com-08253720
[Jul 2 15:57:13] WARNING[2509]: chan_sip.c:1949 retrans_pkt: Maximum retries exceeded on transmission dce84d8-16216490-13c4-486ba4ce-2d0d42e8-13f7@wapp.com for seqno 1 (Critical Response)
[Jul 2 15:57:13] WARNING[2509]: chan_sip.c:1973 retrans_pkt: Hanging up call dce84d8-16216490-13c4-486ba4ce-2d0d42e8-13f7@wapp.com - no reply to our critical packet.
-- Hungup 'IAX2/trunk_1-4'
== Spawn extension (default, 1513203, 1) exited non-zero on 'SIP/wapp.com-08253720'
(the extra “15” on the initial dialled number is due to the routes of the Nortel - it is stripped off before sending to the second Asterisk box)
Both the asterisk and Nortel boxes are on the same network leg, so there is no NAT involved there.
Any ideas?