SIP not hanging up

Asterisk Asterisk Support
1

I’m slowly integrating an Asterisk system into an Iwatsu ECS system over SIP extensions (to and from Iwatsu). Most of it is working except for a small hang up problem. The system is is connected PSTN <–> Iwatsu <–> Asterisk. Both Iwatsu and Asterisk have phones connected to them and I can dial extensions between the two systems. After hanging up on a phone on the Asterisk end the Iwatsu end doesn’t get a hang up signal and just holds the line forever this can be either from calling an extension or calling an outside line through Iwatsu. Any ideas?

Thanks,
-dustin

2

Your best bet is to debug the SIP session, you should see a BYE from Asterisk when the call hangs up. It is most likely a problem with your other PBX if Asterisk is sending the correct SIP messages.

3

Ok I turned full debugging on and I found a few SIP “BYE” message:

<------------->
[Apr 10 08:23:23] VERBOSE[2824] logger.c: — (12 headers 0 lines) —
[Apr 10 08:23:26] VERBOSE[2824] logger.c:
<— SIP read from 192.168.1.55:5060 —>
BYE sip:233@192.168.1.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55;branch=z9hG4bKd3c98bfeFD594E5D
From: “602” sip:602@192.168.1.40;tag=1B7A2266-85225B1F
To: sip:233@192.168.1.40;user=phone;tag=as63beda70
CSeq: 3 BYE
Call-ID: f17cb9da-5969ab70-e13c6f99@192.168.1.55
Contact: sip:602@192.168.1.55
User-Agent: PolycomSoundPointIP-SPIP_301-UA/2.0.1.0291
Proxy-Authorization: Digest username=“602”, realm=“asterisk”, nonce=“2e13a7be”, uri=“sip:233@192.168.1.40:5060;user=phone”, response=“f8c36817159b95f42985053d6f80b280”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0

I’m curious is this stating that a BYE was sent from 602@192.168.1.40 to 233@192.168.1.40? If so I don’t understand because 233@192.168.1.40 does not exist…233@iwatsu does…

Then after I hung up the phone that wouldn’t hang up I got:

<------------->
[Apr 10 08:26:05] VERBOSE[2824] logger.c: — (10 headers 10 lines) —
[Apr 10 08:26:05] VERBOSE[2824] logger.c:
<— Transmitting (no NAT) to 192.168.1.3:5060 —>
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-461b9051-2a0ec4b2-5164;received=192.168.1.3
From: sip:233@iwatsu;tag=301a8c0-13c4-461b8fb0-2a0c4dea-4c1b
To: "602"sip:602@192.168.1.40;tag=as2b429e68
Call-ID: 6407329472d76cef7addba2f5e4bb968@192.168.1.40
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

So it looks like the asterisk system does know to route the BYE signal to @iwatsu instead of itself…any ideas?

Thanks,
-dustin