AA Transfers Failing

[pcamera254]

Got a question, alright so first of the lay of the land. I have a DID, first hop inside my network from my sip proxy is my firewall. Not seeing anything blocking in the firewall, after that I go to my Voice router 2811 acting as a cube. Registered to my sip provider and calls inbound are working 4444 is the destination pattern inside my network. Next in the path is my sip trunk to the CUCM. From their I have a CTI Route Point @ 4444, then I have a Sip trunk to my CUC and a destination pattern pointing to that trunk. The number to be found is 2900 Its my Voicemail Pilot. Now on to the CUC I have a opening greeting, options xy and z. Call’s make it all the way to this part and continue on after a user selects an option,  well kinda. So I select option 1 and it supposed to connect me to extension 5000 it attempts to but then goes to Voicemail (When Im calling internally) When im calling externally (PSTN) I just get silence it attempts the transfer but nothing happens no phones ring. I did my research and it was recommended I Setup transfer rules for users this is already done but the outcome is the same Internally I get voicemail's Externally I get silence. It's like calls are not being handed off the the CUCM like they should. If that were the case though then my Meet me with authorization would fail before connect right? Before you answer that let me describe this part, Host opens a meet me bridge @ 2220 end users call 2222 from their they are prompted to say their name. Next the host authorizes them, All of this is working the problem with this is when the end user joins the bridge from AA it immediately drops off the conference. Anyone Have any ideas where I can focus my efforts?

Im using 9.1 outbound isn't working but that's irrelevant. I have checked my CSS and RP everything is in the same CSS and RP. I know Im probably doing it wrong but Im learning any input is welcome. 

[jagpreet89]

Alright. You have put in a lot of information here, which is good. Although, the formatting could have been better, just a note.

Lets divide our task at hand and then conquer:

> Issue one: Internal calls to 4444

      ++ Call Flow: Cisco IP Phone >> Dials 4444 >> SIP Trunk >> CUC.

      ++ User selects Option 1, the call should transfer to Cisco IP Phone with extension 5000, correct?

      ++ But it goes to a Voicemail. Is this voicemail of the user with the extension 5000 or is this the Generic CUC greeting?

Please answer the above and lets fix this first and I will then list the other issues. If that works for you, let me know.

Regards,

Jagpreet

[pcamera254]

Call Flow is accurate

Correct

This is the users voicemail box also I am not getting any mwi's on my devices as well as no messages are making it to the end user voicemail box. If I go from 5080 >>> 5000 I can leave a message as expected its just when I go to the call handler @ the CUC .

Thank you 

Phillip 

[jagpreet89 · Accepted solution]

Phillip,

What Unity is doing, it sends you to the voicemail box 5000 instead of sending you to the CUCM IP Phone extension since for it 5000 is a voicemail box extension.

A good setup would use a mask for the voicemail boxes, something like:

        > If your DID is 555-777-5000

        > Your voicemail mask could be 777XXXX.

If your voicemail boxes and your IP Phone extensions are exactly the same, you will run into such issues.

Regards,

Jagpreet

[pcamera254]

Outstanding amazing what 4 little X's can do, that fixes the Both problems. Thank you so much Learning curve on the CUC is... Hefty Thanks again