10-06-2014 08:28 AM - edited 03-17-2019 12:26 AM
Hi,
we have a 3rd Party Attendant Console connected with a SIP Trunk to CUCM with a Route Pattern pointed to SIP Trunk.
Here is first the Call Flow:
1. External/Internal call to main number
2. Attandant Console Software and Cisco Phone Ring and the call is not be answered
3. Another Cisco Phone gets Call Pickup Notification and push Pickup
4. The External/Internal hear Queue music from Attendant Console and the 2nd Cisco Phone rings
5. 2nd Phone Answer now the Call and the external/internal Call Disconnects
6. 2nd Cisco Phone looks like its connected and time is counting
From the SIP trace of the 3rd Party Attendant Console i see a BYE Message from the Route Pattern Extension to the EXTERNAL/INTERNAL Caller and thats the reason why the 2nd Phone and the Caller are not connected.
Anyone have same behavior or similar?
Here is the BYE Message:
516 is the SIP Trunk RP and the Outbound Extension to the CUCM from Attendant Console.
976 is the caller
If i click the PickUp Button on the 2nd Phone i see the following in Trace:
SIP/2.0 200 OK
From: <sip:956@192.168.50.5>;tag=5571139~5150ddbf-e085-4401-91b4-6e12bdd9e389-33333940
To: "Andrew Riddick" <sip:976@192.168.50.10>;tag=CYkcysZL
Call-ID: fe2980bffc514e1a906da08918ce580a
CSeq: 101 UPDATE
Contact: "Operator" <sip:976@192.168.50.10;transport=tcp>
Max-Forwards: 70
P-Asserted-Identity: "Andrew Riddick" <sip:976@192.168.50.10>
Remote-Party-Id: "Andrew Riddick" <sip:976@192.168.50.10>;party=calling;privacy=off;screen=no
User-Agent: (samwin core service 6.3.0.5)
Via: SIP/2.0/TCP 192.168.50.5:5060;branch=z9hG4bK1ee494945ad00
Content-Length: 0
UPDATE sip:976@192.168.50.10;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.50.5:5060;branch=z9hG4bK1ee494945ad00
From: <sip:956@192.168.50.5>;tag=5571139~5150ddbf-e085-4401-91b4-6e12bdd9e389-33333940
To: "Andrew Riddick" <sip:976@192.168.50.10>;tag=CYkcysZL
Date: Mon, 06 Oct 2014 15:17:04 GMT
Call-ID: fe2980bffc514e1a906da08918ce580a
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
Supported: timer,resource-priority,replaces
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 UPDATE
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Min-SE: 500
P-Asserted-Identity: "LAB Test Pickup" <sip:957@192.168.50.5>
Remote-Party-ID: "LAB Test Pickup" <sip:957@192.168.50.5>;party=calling;screen=yes;privacy=off
Contact: <sip:956@192.168.50.5:5060;transport=tcp>
Content-Length: 0
956 is Attendant Console Cisco Phone
957 Cisco Phone with same Pickup Group as 956
Maybe someone has a hint.
10-07-2014 04:33 AM
This is most likely an MTP issue. The reason you are experiencing this is due to the break in media transimission when the call is put on hold for subsequent transfer to the second phone that has pressed the pick up soft key. During this process the call is "inactive" in sip terms and some 3rd partys do not deal well with this. It is advisable to enable MTP in sip trunks to 3rd party devices..
10-07-2014 04:46 AM
I tried it with and without MTP. I thinnk the BYE message from operator (3rd party) is the problem. But i could not solve it by now.
10-07-2014 04:50 AM
Please enable MTP required on the sip trunk. Reset the sip trunk. Make sure you have an MTP device available in the MRG of the sip trunk and your phones.
do another test call and send me the CUCM traces (use RTMT) to download cucm traces and send ocer, Include the calling and called number and time of call
https://supportforums.cisco.com/document/126666/collecting-cucm-traces-cucm-862-tac-sr
10-09-2014 02:55 AM
Hi Ayodeji,
thx. I have now the confirmation from 3rd party company that it's a problem from them.
thx buddy
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