This is a little curly.
I have two asterisk boxes.
One is on a public IP, in a data centre, which is used for external handsets. PBX-1
The second is in the office behind a SIP ignorant NAT router. PBX-2
The two systems work flawlessly on their own.
I have connected them together with an IAX2 trunk whereby PBX-2 registers to PBX-1
This allows us to call from a SIP extension on one PBX to a SIP extension on the other and the call quality is fine. Caller ID works. Life is sweet…except for one annoyance they want me to fix.
If I call from extension 101 or PBX-1 to 202 on PBX-2 and hand up before they answer, extension 202 continues to ring.
And vice versa. Furthermore, the call on the handset stays active for 1-3 secs after the other party hangs up. Not a deal breaker, but a deal spoiler.
This doesnt seem to happen with a SIP trunk, however there are complications in using SIP especially in relation to multiple registrations behind NAT from a single IP to an Asterisk box. Specifically, regardless of the context=wherever that is in your SIP peer, you always end up with inbound calls starting in the context of the last SIP peer you registered from that IP.
I am running an older version, 1.4.22, becuase at the time it seemed to work the best with the SIP providers we have. It has nothing to do with system load as it does this when this is the only call.
Can anyone shed any light on this one
I would like it to work with IAX, however if I am forced back to using SIP, I’ll have to do some major rework of our dial plans as we have multiple contexts for multiple departments / tenants.