Trunk Anti Tromboning Feature doesn't work on SIP trunk if MUSIC or RAN is given

[atmaal5]

Hi,

CS1000M cpp4 in Release 7.5 connected to AAEP Avaya via SIP trunk.
A call to AAEP is done via SIP, AAEP presents a voice menu, the caller dials a digit and AAEP initiates
a transfer to an extension in CS1000M by sending a REFER to SIP Gateway.
If there is no MUSIC or RAN and so the call is routed directly to a set, as soons as the set answers the call the 2 SIP trunks are released and the call is done internally in CS1000M : this is TAT feature.
If there is a waiting message RAN MUSIC, the TAT feature never works.
Facility (TAT ROSE encoding) messages are exchange to trigger the TAT but CS1000M return a RESULT ERROR.
Does anybody experience a similar problem?
Thks,

 

[intelligentium]

Found the below post somewhere
"TRO/TAT are MCDN features and currently only supported between two CS 1K nodes that understand the tunnelled MCDN messages on SIP. This feature does not work with any other nortel (including ICP) or third party nodes that do understand MCDN (Note the media while using SIP IP peer is always optimized ).
TRO feature doesn't work with any 3rd party switch (including Nortel IVR [MPS/ICP]), It works with switches which understands proprietary messages from/to M1/CS1K.

In a nut shell, CS1K does not support SIP trunk optimization with any third party that transfers the call back to CS 1000 using REFER."
---
Well anyways can you post the dch mon 2 facility messages that you have mentioned giving an error. Not sure if i can help. But can have a look at it.

 

[intelligentium]

Atmaal,
Can you describe the scenario a little bit. When is the music wait being played.
If possible can you give a callflow skeleton diagram.
Which release is the system.

 

[atmaal5]

Thanks for your reply.
I already red the comment you post on Avaya site but the it is really stupid: MCDN feature are local between the SIP Gateway in signalling server and the Dchannel IP on CS1000M : in my case it is not an End to END MCDN feature encapsulated in SIP messsage for example. When a REFER SIP message is sent by AAEP( an Avaya product to manage messages and call scripting) an 2nd trunk is taken and when the called set answer the TAT feature works fine and the 2 SIP trunks are released ( Facility TAT ROSE messages are sent by the SIP Gateway).
Scenario :
Incoming PSTN ---routed to AAEP via SIP trunk-----AAEP plays meessage and the caller dials 1 for example
------AAEP initiate a transfer to a CDN number for example controlled by symposium ----if no agent available , a patient message is played by a RAN for example.
At this time a connect is sent to SIP gateway, Facility are exchanged but TAT feature doesn't work.


My English is bad,sorry for that

Here is the Dchannel traces with my comments (Dchannel IP) :


A MUSI or RAN is given to the caller so CONNECT

DCH 20 OMSG CONNECT REF 00008181 CH 252 0 0 0 TOD 11:05:04 CK 1D00EE95
HEADER:01 14 00 05 08 02 81 81 07

DCH 20 IMSG CONN ACK REF 00008181 CH 252 0 0 0 TOD 11:05:04 CK 1D00EEB6
HEADER:2F 14 00 05 08 02 01 81 0F

A ROSE TAT INVOKE is sent bye the sip Gateway

DCH 20 IMSG FACILITY REF 00000234 CH 252 0 5 19 TOD 11:05:04 CK 1D00EECD
HEADER:2F 14 00 1C 08 02 82 34 62
FAC :1C 15 11 0F 01 11 02 01 02 02 01 02 30 09
80 04 00 00 81 81 81 01 01
comment :
11 ROSE
OF TAT feature
01 INVOKE
11 length
02 01 02 Invoke ID (TLV) 02 TAT 3 Party Invoke ID
02 01 02 Operation value tag 02 TAT 3 Party Interaction
30 09 80 04 etc..Call reference and 81 01 01 TAT redirection Reason Tag 01 TAT call Network Transfer

DCH 20 OMSG FACILITY REF 00000234 CH 252 0 5 19 TOD 11:05:04 CK 1D00EED4
HEADER:01 14 00 11 08 02 02 34 62
FAC :1C 0A 11 BE A3 06 02 01 02 02 01 13

comment :
A3 for me means Return ERROR but i have no idea about the 02 01 13 (13?????? error cause)

DCH 20 IMSG FACILITY REF 00008181 CH 252 0 0 0 TOD 11:05:04 CK 1D00EEDD
HEADER:2F 14 00 11 08 02 01 81 62
FAC :1C 0A 11 0F 03 06 02 01 02 02 01 13