CUE doesn't record messages from external calls.

[douglasclark]

I have a 2811 running CME 4.0(1) (yes I know it's old) with a AIM-CUE module running 3.2.3 (the reason for the ancient CME.

After much mucking about I have managed to get the CME to forward unanswered calls to the CUE, which is responding with the correct mailbox greeting for each DID number called. The problem is that the CUE is not recording the messages left by external callers. I can see this as the message count on the CUE is not incrementing (show voicemail mailboxes) when i leave a test message.

On the CME under voice service voip i have the following setup

allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 no supplementary-service h450.2
 no supplementary-service h450.3
 supplementary-service h450.12
 no supplementary-service sip moved-temporarily

What have I missed? I had assumed that once I had concatenated the SIP trunks through the CME to CUE the RTP traffic would also follow. I assume that's what's causing the voice message not to be recorded by the CUE.

Thanks in advance for any help.

[Nadeem Ahmed]

Hello Douglas,

Couple of thing need to be checked

1) what happened when  you try to leave message internally?

2) what happened when you try to leave message externally ? do you get any announcement being played by CUE something like opening greeting rather than user personal greeting?

3) What is extension range you have configured is it 4 digit or 3 digit?

3) can  you please collect the debug voice ccapi inout and debug ccsip messages on CME.

Regards

Nadeem Ahmed

[douglasclark]

1) Calling from an internal extension I get the correct greeting (from the GDM) and i can see on the CUE that there is a message on the system. I don't however have a MWI on the IP phone (I'll deal with that later) 

2) External call I get the correct mailbox greeting associated with the DID I'm dialing

3)I'm using 4 digit extensions. Does this make any difference?

4) Don't have access at the moment to grab some debug as I'm out of the office. I'll post when i can.

[douglasclark]

Find attached sanitised debug output as requested.

[Nadeem Ahmed]

Debug ccsip message is missing. yes it could be an issue with external caller when they are try to leave the message. are they getting VM tone leave message after beep etc... and does CUE play any kind of message msg delivered.

Br,

Nadeem

[douglasclark]

Sorry didn't see your request for ccsip debug, I have attached it.

When a caller gets direct to the CUE you get the mailbox greeting, ten then "leave a message after the tone" then the tone.

No message deliver message is played. I usually hang up to terminate the call.

[Nadeem Ahmed]

What is the dial-peer configuration which is pointing to CUE as SIP trunk. Is Codec G711 is hard coded if not you can please try and make one test from externally.

Dial-peer voice 11 voip =====Pointing to CUE.

codec g711ulaw

Meanwhile i will go through the traces 

Br,

Nadeem

[douglasclark]

The dial-peer is set to hard code to g711. I've included the complete config below. Thanks for taking a look at the debug.

dial-peer voice 8888 voip
 description Voicemail
 destination-pattern 8888
 session protocol sipv2
 session target ipv4:192.168.1.199
 dtmf-relay sip-notify
 codec g711ulaw
 no vad

[douglasclark]

After some more investigation on the CUE i seem to have found that the RTP traffic is not getting to the CUE given the trace messages below,

4868 06/04 12:10:51.430 ACCN LMED 0 Recording Started
4868 06/04 12:10:51.430 ACCN VBRW 0 Task:16000000014 WFDTMFDialogServicesAdapterImpl: RecState: terminationChars: #
4868 06/04 12:10:51.430 ACCN VBRW 0 Task:16000000014 WFDTMFDialogServicesAdapterImpl:  ENTERING RecState.start() aborted=false
4864 06/04 12:10:51.430 ACCN LMED 0 [LIB_MEDIA_RTP_SENDER] RTPReceiver::run started1 Thread:com.cisco.wf.media.RTPReceiver@5cde5cde
4864 06/04 12:10:51.431 ACCN LMED 0 Enter: DoubleBufferDatagramInputStream:read(Buffer), buf:Income-1@DATAI.2
12335 06/04 12:10:51.832 ACCN LMED 0 [RTPReceiverRAB-4] java.net.SocketTimeoutException: The operation timed out
12335 06/04 12:10:51.832 ACCN LMED 0 Enter: DoubleBufferDatagramInputStream:read(byte[] buf
12335 06/04 12:10:52.332 ACCN LMED 0 [RTPReceiverRAB-4] java.net.SocketTimeoutException: The operation timed out
12335 06/04 12:10:52.333 ACCN LMED 0 Enter: DoubleBufferDatagramInputStream:read(byte[] buf
12335 06/04 12:10:52.833 ACCN LMED 0 [RTPReceiverRAB-4] java.net.SocketTimeoutException: The operation timed out
12335 06/04 12:10:52.833 ACCN LMED 0 Enter: DoubleBufferDatagramInputStream:read(byte[] buf
12335 06/04 12:10:53.333 ACCN LMED 0 [RTPReceiverRAB-4] java.net.SocketTimeoutException: The operation timed out
12335 06/04 12:10:53.333 ACCN LMED 0 Enter: DoubleBufferDatagramInputStream:read(byte[] buf

I've done the same trace on an internal call that is being recorded and that looks fine. So appears that the RTP stream from the SIP SP and the CUE is broken, and I'm guessing that this is because the CUE is a private address that the SP cannot route to.

Any ideas how to get around this?