Speed Dials not working after moving to SIP

[sanmar]

I moved to SIP this week and I had some speed dials set up and now when we go to use them, they just ring and don't go to voicemail when they should. Anyone have any ideas. Of course I'm waiting to hear back from Mitel. Thanks

[johnp]

Not much detail as to what the speed calls do and what you mean moved to sip

[sanmar]

Speed calls usually just dial out of system.  I have my helpdesk set up as one. During the week the DID (which I have set up as a speed call) forwards to an ACD path and on the weekends, it forwards to their cell phone so they can work from home. I moved from PRI Trunks to SIP on Wednesday and ever since then the speed dial or (outbound forwarding does not work). It connects the call and then drops it within 8 seconds. Nothing has changed on the PBX other than moving the lines to SIP.

[acejavelin]

OK, so the speed dial works, you said "it connects the call and drops it within 8 seconds", so the speed dial code itself works from your statement.  I assume you tried just dialing the speed dial code from your phone to verify? And I assume the ACD path portion works fine?

It sound like the problem is taking an incoming SIP call and redirecting it back out as another SIP call unsupervised? There are numerous reasons this could occur, COR, COS, SIP Peer Profile settings, incorrect outbound CallerID, etc. Does a supervised transfer work? Meaning, if you call your DID number, answer it, then try to transfer it to another number or speed call entry, does the transfer start successfully, can you communicate with the far-end, press Trans/Conf again and make a 3-way call, disconnect and the call stays up?

How are calls triggered to go to the cell phone, when everyone is logged out of the ACD path, when the system is in Night service, via a Nupoint Call Director mailbox, or some other means?

Often times the best way to troubleshoot this is to make test calls and analyse the SIP messaging to see where the call is being disconnected and why...

[sarond]

What is your system type?
I think this belongs in one of the system forums as it sound like it is to do with SIP Trunks.

[sanmar]

Supervised transfers all work. I still have my 800#s on my PRI and when you call them and dial the DID the forwarding works. It just doesn't when you dial the DID (which is on the SIP). The calls are triggered to cell when I change the point to number on the speed dial. I change from the path to a direct cell #. I ran a trace route from the MBG and made test calls but the "bye" request I get from the SIP ends up in different places, sometimes in the PBX and sometimes in the MBG. The system I have is a 3300.

[Dutch]

As Ace indicated migrating to SIP needs a couple of variables before it works correctly. So without looking to many details (and thus making many assumptions unfortunately)

If you answer a DID call from SIP to internal Phone can you maintain a call for 30 mins.? Its usually a test I do to see if the session timer works correctly. Aside from that if signaling is fine but RTP doesnt work fine I start looking at the SDP options as well. Conferencing and Hold retrieve from hold usually give a quick indicatation if invites and Re-invites work correctly.

Usually after confirming that all the basic functions work I would look into the routing you need.

Usually with no RTP in my experience in most cases I have look at:

SIP Peer profile SDP Options
Avoid Signaling Hold to the Peer - Yes
Suppress Use of SDP Inactive Media Streams - Yes
Prevent the Use of IP Address 0.0.0.0 in SDP Messages - Yes

Also if your provider supports PRACK enable it on MBG and Mitel.

I hope this info helps a bit but your description of the issue can be in a lot of things and putting them down all means a wall of tekst.

In my experience the best approach is to use a certified trunk. In that case the settings are qualified for various uses and available via the Mitel website and/or provider of your SIP trunk. Using those settings should result in a working setup.

Cheers,

Dutch

[sanmar]

Problem solved, just wanted to share with everyone. in SIP Peer Profile under SDP Options, "Renegotiate SDP to enforce Symmetric Codec" needs to be set to YES.  Mitel will be updating their white paper on this.  Thanks everyone for your input.