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Greetings forum,

Please save me from offing myself with one of several close-at-hand phone cords. Even fond memories of taking Ted's class may not get me through this difficult time.

I have constructed a CME/CUE demonstration lab with the following equipment:

Latest release version of CME on a 2801 router. Matching release version of CUE on an CUE-AIM module.
Cisco 7965G phone
Cisco 7941G phone

We are using the SCCP protocol with these phones with no problem. We even did extension mobility and that works great.

There has been great resistance by one of my potential customers to the high price of Cisco phones. I investigated some alternatives and decided to try 2 Linksys phones with the system(formerly sippura). I also used the X-Lite SIP phone to reproduce the same problem. Here is the MAJOR problem I am unable to solve:

If SIP phone is registered:

Call placed from outside (a H.323 trunk) works fine. On no-answer, rolls over to voicemail.
Call placed from inside (SCCP phone) works fine. On no-answer, rolls over to voicemail.

I can see dynamically created dial-peers present in the dialplan. If I debug the SIP messages on the CME box, I see the first call attempt to call the sip phone. After the timeout I see a second call the CME box places to the CUE module. On this call there is a "Diversion: 4001" (4001 = SIP phone's extension) header. This notifies the CUE that the original call was placed to 4001 and it goes to the user's mailbox appropriately. So far so good.....

If the SIP phone is NOT registered:

Calls placed from the outside (a H.323 trunk) to the SIP phone's prior extension roll over to voicemail, where it promptly asks me to enter my ID. If I debug the SIP messages, there is no original call placed. The first call is directly to VM. Unfortunately, this SIP message does not contain the "Diversion: 4001" header?! I suppose since there was never technically a "diversion", that it doesnt think it needs this header, but without it, how is the CUE supposed to figure out what mailbox to go to?
Calls placed from the inside (a SCCP Cisco phone) immediately get a fast busy - "Invalid Number". This is backed up by the fact that the dynamic dial-peer entry for my SIP extension has been revoked. This is not desirable behavior either.

For outside calls (via H.323 trunk) As long as I have created a "number X dn X" command under any voice register pool, the system appears to know that this is a valid extension that can be sent to VM (CUE), but it doesnt send anything along with it to tell CUE what mailbox is being sent to. This is PROBLEM #1.

For inside calls (Via SSCP phones), the "number X dn X" under the voice register pool has no effect. If the phone isn't registered, the dynamic dial-peer isn't present and the number is INVALID. This is PROBLEM #2.

I can't explain the inconsistant behavior between problem #1 and #2, nor do I have any clue how I can get the "Diversion" header on those SIP messages to solve problem #1 in the first place. Thanks in advance for any help.