Strange issue with interop between Cisco CCM and 3300

[1quikst162]

We're on 5.0 SP2 PR2. The CCM is 7.0 (Cisco Call Manager).

I have a SIP trunk established between a centralized 3300 and the CCM Publisher. All other 3300's are sending the digits via ARS digits dialed to the centralized 3300 and then out the SIP trunk to the CCM.

When it was established, we tested and everything was great. I call them, they call us, we both heard each other. Lately (as in 3 days later) after a while no audio was passing. I call over there, nothing. They call here, nothing. Forward calls to voicemail and still same problem. No message is left on their side and they don't hear our voicemail system.

Kinda at a loss as to what broke. There have been no chances to the SIP trunk or the centralized controller on the 3300 side. I saw the blurbs about SDP dupliacte offers, made the mods and still no audio. Session timer is set to Zero (disabled). I had it at 90 when it was working. I bumped it up to 1800 per an older Cisco <> Mitel document I found to no avail. I'm at a loss as to what happened.  Any insight?

I heard rumors there's an updated MOL document that outlines how to configure a SIP trunk from a 3300 to CCM. I found a Cisco document that I followed that didn't really match up to what we have (it was on the old 7.1 release).

[martyn]

If you export the SIP trunk traffic from the 3300 and the CUBE to wiresharsk, does it show that there is traffic passing between the two, if so does it look correct? Also are there any alarms showing on the 3300 relating to the SIP trunk?

[1quikst162]

If you export the SIP trunk traffic from the 3300 and the CUBE to wiresharsk, does it show that there is traffic passing between the two, if so does it look correct? Also are there any alarms showing on the 3300 relating to the SIP trunk?

I'll have to try that in the morning, I'm having some other issues right now (other site hard down, need to divert attention over there first).  There are no alarms on the system that reference this specific SIP trunk.

In the morning I'll make a call to there and SIP trace. Could you provide an example of what SIP traffic should look like?

[martyn]

You should see a 100 trying, 180 ringing, and a 200 ok if the call is successful. If not then you will probably see 40x errors that will give a hint as to what is going on.

[1quikst162]

Thanks. I haven't troubleshooted VOIP issues for a while, bit rusty in the call setup & teardown procedures

Would you know offhand if a MOL document exists for a 5.0+ MCD release to connect to a Cisco enviroment?

[martyn]

There is only an old document available from what I can see. This should still apply though.

Sorry, should have clarified, is the SIP trunk to CUCM or to a CUBE on the cisco side?

[1quikst162]

I used that older document (like MCD 7.1 or something).

I used that as the guideline, modifying accordingly (because of course, MCD 7.1 to MCD 5.0 SP2 PR2 is quite a bit of changes).

I'm not sure what you're talking about the CUBE/Cisco thing (I have honestly never touched one of these enviroments, only dreamed of it). The SIP trunk is connected to the Publisher on their side. That's the extent of the Cisco enviorment that I know of (I'm hoping to get some diagrams/etc of how their VOIP is so I can understand better).

[1quikst162]

So here's something strange.

I get home last night night around 2am.  Only changes I made were to disable the session timer (I backed out everything else). The SIP peer is basically the default when you add one. I busy'd and RTS'd the sip peer to bounce it to apply changes. No audio.

This morning on the way in, I get a call from the Cisco office and I can hear him just fine. I was actually on my twinned number. Call came in the SIP trunk, routed over IP/XNET trunk to another 3300, then out the PRI to my cell phone. No problems. I called him back from the office (so ip phone --> controller --> ip/xnet --> sip --> cisco) and audio was fine.

/completely confused why it didn't work last night.

[martyn]

Someone must have been fiddling. Things don't just start stop and start working like that without some sort of intervention. 

[LoopyLou]

You look after the Cisco too?

[martyn]

I'm not sure what you're talking about the CUBE/Cisco thing (I have honestly never touched one of these enviroments, only dreamed of it). The SIP trunk is connected to the Publisher on their side. That's the extent of the Cisco enviorment that I know of (I'm hoping to get some diagrams/etc of how their VOIP is so I can understand better).

Sorry forgot to follow this bit up.
The CUBE (Cisco Unified Border Element) is the ISR G2 router (usually 2900's on newer systems), and is where the SIP session terminates, and you can apply profiles, etc to modify, or basically manipulate what ever you need to in SIP for interop between two systems.
Generally you would have a SIP trunk (or call leg) between CUCM and the CUBE, and then another between the CUBE and the TSP, or other systems (ie 3300). You can just have the SIP trunk between two systems (ie CUCM Pub/sub to 3300), but a CUBE is a a lot more flexible in manipulating as well as obscuring parts of the SIP header, depending on the requirements.