CUC Unable to transfer to an external number after Caller Input

[Joseph Rivera-Santiago]

We have a set of call handlers that direct calls based on caller input, i.e.:

1. Call external # for Office
2. Greeting plays
3. Caller chooses Option 1
4. Gets transferred to 2nd level call handler
5. Caller chooses Option 6
6. It then should get transferred to an external contact number

-Instead it plays the transfer greeting, "Please wait while I transfer your call", MOH plays, and then dead silence.

It does not ring, then you are forced to hang it up.  It's like the call gets sent (to the SIP router I am guessing), but never gets there or gets terminated on that end.

All other options within the handlers work (transferring to an internal extension, and transfer to an additional call handler)

Any help would be greatly appreciated, or any ideas on where else to troubleshoot.

[Vivek Batra]

Ok, mid night here. Update the result and will catch you tomorrow.

- Vivek

[Joseph Rivera-Santiago]

Yea still running into the same issue.

Guess I need to understand it better.  Once Unity handles the call, does it still need to go to CUCM to then be sent to the SIP router to then complete the call?

Would it have anything with the Voicemail pilot?

[Vivek Batra]

You definitely need CUCM for CUC to process any call. CUC can't live without call agent.

Your call flow is like this;

ISP (SIP) -> Gateway -> H.323 -> CUCM -> CUC -> CUCM ->  H.323 -> Gateway -> ISP (SIP)

We need to resolve it step by step. As per our last discussions, can is hitting CUC and we need to take it further to CUCM by making relevant changes there viz RP, Partition, CSS etc.

- Vivek

[Joseph Rivera-Santiago]

Ah ok, is there anything I should be checking on the CUC side then?

It does seem to be making it out of CUC, into CUCM, grabbing the right pattern etc.

I just can't seem to find where it is breaking or where else to troubleshoot

Thanks again for all the help

[ciscoceci]

Hello Joseph,

could you try transfer yor call to internal extension and configured this extension with his fwd to external number??. with it you test if exist a permissions problem (CSS, PTT) betwen CUCM and CUC

Regards Anna

[Joseph Rivera-Santiago]

Oh awesome idea, ok let me set that up and I'll report back

[Joseph Rivera-Santiago]

Wow ok that works, that went through all the way to the external number.

But I have no idea why, what would be the next step?

I have a 5 Call Handler setup, and they have extensions assigned in CUC.  However is there something else that needs to be created on the CUCM side?  CTI Port or Route Point?

Thanks again for all the help

[Joseph Rivera-Santiago]

Any other for this?  I forwarded to an extension that is attached to a phone, and that worked.

Could I create a new DN and attach it to a CTI Port / Route Point and forward the call that way?

[Vivek Batra]

Can you please clarify under option 6 in AA, you've configured external number in CUC (Call Action -> Transfer to Alternate Contact Number) OR expecting caller to enter the external number manually.

On the basis of configuration, we will check further.

- Vivek

[Joseph Rivera-Santiago]

I have set it back to the external number yes.  I was trying a few other workarounds, mainly forwarding to a DN assigned to a CTI Port and then forwarding all to that external number.  Which was working, and then unfortunately stopped for no apparent reason.

I have set Option 6 from the 1st menu back to the external number.

Thanks

[Vivek Batra]

If you mean you've configured alternate contact number under caller input, then first thing here no restrictions table is applied here and call must go to CUCM.

What I can suggest you to have new partition and CSS and assign this new CSS to voicemail port. This CSS should see only newly created partition associated with number (RP) to be dialed. Ensure this RP has correct RL assigned. Then check in gateway if this call is reaching there or not.

- Vivek

Please rate useful posts.

[Joseph Rivera-Santiago]

Alright I created a new Partitiion/CSS/Pattern and assigned to correct RL containing our SIP router.

I also change an existing vmail port to use those newly created settings.

What would be the easiest way to check if it is hitting the gateway?

debug ccsip all , messages?

[Vivek Batra]

You might be having multiple voicemail ports and not sure which will be used...better to use new CSS on all ports.

Use debug voice ccapi inout and debug voip dialpeer inout

-Vivek

[Joseph Rivera-Santiago]

It's looking like we are having performance issues on the SIP side.

When I call from my direct extension, the call goes through, and I can see it hit SIP.

When I try from Unity (AA/IVR etc), the call does not go through, and does not show up in SIP.

When I check logs, I am getting quite a few:

%VOICE_IEC-3-GW: H323: Internal Error (No answer from user): IEC=1.1.19.5.23.0 on callID 4215

%VOICE_IEC-3-GW: C SCRIPTS: Internal Error (CPU high): IEC=1.1.181.11.3.0 on callID 5207

%IVR-3-LOW_CPU_RESOURCE: IVR: System experiencing high cpu utilization (96/100).
     Call (callID=5207) is rejected.

It's an older 2851 still running 12.2 unfortunately

[lorenz84gDD]

do you have a dial-peer on your router for the destination where the call is going to be transferred?