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Beginner

RDP transfer fail

When user tries to transfer an answered call to a RDP enabled directory number, he is unable to transfer the call.  
Scenario:
Customer dials User A - User A answers - User A dials User B- User B has enabled RDP to his mobile number - User B answers in mobile phone - Customer is on hold - User A tries to transfer the call to User B - call to User B drops out - Only Customer and user A stays connected.

 

Any suggestions?

 

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3 REPLIES 3
Participant

Re: RDP transfer fail

So basically the call drops when an external number calls an IP Phone and IP Phone tries to do a consult transfer to another external number?. Is this a SIP trunk through which both the legs establish, if so do you by any chance have Media Termination Point Required checked on that trunk?
Also please provide the complete call flow (eg ITSP > SIP > CUBE > SIP > CUCM > SIP > CUBE > SIP> ITSP) along with detailed Callmanager traces for a failed call.

https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200787-How-to-Collect-Traces-for-CUCM-9-x-10-x.html

Please rate if you find this helpful.

Regards,
Aeby



Please rate if you find this helpful.

Regards,
Aeby
Beginner

Re: RDP transfer fail

MTP isn't checked, which I confirmed. The call flow is ITSP > SIP > CUBE > SIP > CUCM - extn A > Answers > A Dials extn B > RDP > Mobile > CUCM > SIP > CUBE > ITSP> Call answered by B > A tries to transfer the call to B > Call fails > B disconnected.

When I checked the failed called logs I could see the remote device (Mobile) is sending cancel message followed by 487 request cancelled by CUCM and Bye. Anything I should verify before I enable MTP in sip trunk?

Participant

Re: RDP transfer fail

A cancel is normally send when the session is not established which is not true in our case, you can try enabling MTP and resetting sip trunk to test, but enabling it permanently would mean MTP would be used for every calls over the trunk which would result in more media resource use. If you are enabling it do make sure you have the appropriate resource under the SIP Trunk's MRGL/DP. Do you have this issue if the initial leg is an IP Phone and is not an external number? Also I can take a look at the traces if you can provide them along with the details requested for in my previous post, that should give us more insight on this.

Please rate if you find this helpful.

Regards,
Aeby


Please rate if you find this helpful.

Regards,
Aeby
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