PSTN calls to branch site drops during transfers

[gavenm10240]

I have the following senario:

Incoming Call from PSTN ===BRI===>Voice GW====>WAN===CUCM(HQ)====WAN===>IP Phone A (Branch Site)===TRANSFER==>IP Phone B( same Branch Site)

Call Drops the moment a transfer is selelcted by IP Phone A

Please note this site was woking and suddenly the issue started.

H323 IOS : c2900-universalk9-mz.SPA.151-4.M4.bin

CUCM : 8.6

[Nishant Savalia]

Hi Gaven,

What is the calling number & called number & IP Phone B number?

Also share running config of branch site.

Regards,

Nishant Savalia

[gavenm10240]

Hi Nishant

Phone A : 9111

Phone B : 9110

User from the PSTN dials a ten digit number XXX-XXX-9111==> VGateway==> WAN==CUCM==WAN ===> IP Phone A (9111) //Transfer to IP Phone B (9112) || CALL drops ||

1. Branch site running config is attached as SH RUN on the original mail

[Nishant Savalia]

Hi Gaven

Please check and share the result : -  Initiate call from Internal Extension to Internal Extension and try to transfer internal number itself. See if this is success?

Also Can you please share detailed CUCM trace. 

And how you are transfering the call ? I mean Phone A dials the Phone B number by putting on hold & Phone B rings. Then phone B answers the call and after that Phone A transfer the call. At this point call is being dropped?

Are the phone A & Phone B is in same device pool ?

Regards,

Nishant Savalia

[gavenm10240]

Hi

When IP Phone A calls IP Phone B and transfer to IP phone C, the transfer works.

We have performed the following tests:

Called Number : 011 478 9112

Transfered to    : 011 478 9115

1. Extention 9112 pickup the phone

2. Put us on hold

3. Transfer to extension 9115

4. Extension 9115 answers and accepts the call

5. Extension 9112 hits the transfer button to send the call through.

6. Call cuts, PSTN caller here dead noise.

Below are the debugs performed:

• Debug h225 asn1
• Debug h245 asn1
• Debug isdn q931

 Attached is the output from the debug.

[mohamed helmy]

Hi,

can you please check if the IP Phones use MRGL?

check the region codec if there is any codec mismatch.

also review the MTP of the GW.

Regards,

Mohamed Helmy

[gavenm10240]

Please check and share the result : -  Initiate call from Internal Extension to Internal Extension and try to transfer internal number itself. See if this is success? ===> Calls initiated internally are transfered out succesfully.

Also Can you please share detailed CUCM trace. ===> traces are too big to upload

And how you are transfering the call ? I mean Phone A dials the Phone B number by putting on hold & Phone B rings. Then phone B answers the call and after that Phone A transfer the call. At this point call is being dropped?- Yes, this is how we have been testing the transfer and the call drop the moment the transfer button is initiated for the second time. This is the same instance we are getting a disconnect on the isdn q931 debugs.

Are the phone A & Phone B is in same device pool ?---> All the phones in this branch are in a same device pool.

[gavenm10240]

After reviewing GW Debugs it was see that CallManager is Sending Information message to GW, and when GW forwards this Message to Provider, it disconnect the call.

1198961: Mar 12 12:29:43.391 GMT: H225.0 INCOMING PDU ::=

value H323_UserInformation ::=

    {

      h323-uu-pdu

      {

        h323-message-body information :

        {

          protocolIdentifier { 0 0 8 2250 0 5 }

          callIdentifier

          {

            guid '03B6CDF9A90811E38B2F96531D1C0400'H

          }

        }

        h245Tunneling FALSE

      }

    }

1198962: Mar 12 12:29:43.391 GMT: //27923/03B631D18810/CCAPI/cc_api_call_info:

   Info Digits=, Info Complete=FALSE,

   Interface=0x313B8638, Data Bitmask=0x1, Call Id=27923

1198963: Mar 12 12:29:43.391 GMT: //27922/03B631D18810/CCAPI/ccCallInfo:

   Data Bitmask=0x1, Call Id=27922

1198964: Mar 12 12:29:43.391 GMT: ISDN BR0/2/0 Q931: TX -> INFORMATION pd = 8  callref = 0xDD

1198965: Mar 12 12:29:43.543 GMT: ISDN BR0/2/0 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x5D

                Cause i = 0x82A600000000 - Network out of order

to Stop CUCM from sending this message, we’ve changed the Send H225 User Info Message from “H225 Info for Call Progress Tone” to “Use ANN for Ring Back” Which solved the issue.

[Gmoney]

Hey,Are you using the build-in conference bridge for transfer and conferencing? If so, that ressource is not always available, and that may explain you issue.

Solution:

Create a MTP (Media Termination Point) assign it a MRGL like mohamed.helmy said, and make sure to use that it your device pool configuration, your gateway configuration, and make sure you assign the parameter "default" to the build-in conference bridge for all the phones. (you may use bulk administration to perform that last task)

Regards