Asterisk-3300 SIP call issues

[TechGuru]

I've recently started with a software company and had a 3300 ICP dumped on my lap to get set up/configured so that our developers could write a software interface that will allow our main software product to communicate with Mitel phone systems (via OIG).

We've got most things set up and going, but in testing we're having issues getting calls passed from an Asterisk phone system into  the Mitel system. I'm sure it's something I've missed in SIP configuration, but I don't know what.

The software log file on the phone system is showing the following:

Refused to deliver a event[ServiceFailNotification; 5] without a valid session id

The two systems are connecting, because I can also see:

SIP trunk Aster.42 Account 150 Registrar 192.168.1.42 has registered successfully. The registration will expire in 3600 seconds.

What am I missing to get calls to come in? What needs changed to provide a session id - something on Mitel, or on Asterisk?

Thanks in advance for your time/help/patience.

[martyn]

You don't mention what version of software the system is running, but from the maintenance command can you do a SIP TCPDUMP and then attempt a call, then turn the dump off. The file can be grabbed from the vmail directory via ftp. If you can attach it here then it might provide some more clues

[TechGuru]

My apologies; I should have known well enough to give that info.


MXe-III 3300:
Release level: 7.2 
Active software load: 13.2.0.17


I will get with my testing person today and get back to you with more detailed results/info. Thank you very much!

[lundah]

Mitel has an interop guide on setting this up. The KB article is 06-5160-00002.

[Dogbreath]

sngrep is great for real-time SIP debugging.

[TechGuru]

Thanks for pointing me in the right direction - we've got it working.

I wasn't able to get the KB, but it led me to a few other things that did help. When I tried to get to the KB on Mitel's site, I kept getting an error.