08-10-2011 08:44 PM - edited 03-16-2019 06:24 AM
Hi All,
Im hoping someone can help me out here.PSTN callers are not hearing ringback when tranferred to another extension when call is transported to CUCM via SIP trunk to SIP phones.
PSTN -> 3900 SIP SRST Gateway-> SIP Trunk -> CUCM 7.1.5 -> SIP 7945-> No ringback when transferred to another SIP extension on same cluster.
SIP Trunk in CUCM has MRGL that has ANN assigned to it and MTP checked.SIP profile assigned to the trunk has"Disable early media on 180" checked.
After debug ccsip message, it appears that CUCM isn't sending "100 trying" or "180 Ringing" when transfer is initiated and is going straight into "200 OK". Heres the section of the debug during the transfer.
Aug 11 13:09:02: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
UPDATE sip:0040XXXXXX@172.31.254.67:5060 SIP/2.0
Date: Thu, 11 Aug 2011 03:08:53 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: <sip:71197@10.69.11.1>;tag=6417ce1c-8b93-4ba0-859a-ad892a39e373-44043884
Allow-Events: presence
P-Asserted-Identity: "Ray Choi" <sip:77887@10.69.11.1>
Supported: timer,resource-priority,replaces
Remote-Party-ID: "Ray Choi" <sip:77887@10.69.11.1>;party=calling;screen=yes;privacy=off
Content-Length: 0
User-Agent: Cisco-CUCM7.1
To: <sip:00406313132@172.31.254.67>;tag=3D081A40-1ACD
Contact: <sip:71197@10.69.11.1:5060>
Call-ID: XXXXXXXXXX@172.31.254.67
Via: SIP/2.0/UDP 10.69.11.1:5060;branch=z9hG4bK1dcba7675c7d74
CSeq: 101 UPDATE
Max-Forwards: 70
Aug 11 13:09:02: //261112/11D8A73082F0/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.69.11.1:5060;branch=z9hG4bK1dcba7675c7d74
From: <sip:71197@10.69.11.1>;tag=6417ce1c-8b93-4ba0-859a-ad892a39e373-44043884
To: <sip:004XXXXX@172.31.254.67>;tag=3D081A40-1ACD
Date: Thu, 11 Aug 2011 03:09:02 GMT
Call-ID: XXXXXXXX@172.31.254.67
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 UPDATE
Allow-Events: telephone-event
Contact: <sip:00406313132@172.31.254.67:5060>
Supported: timer
Content-Length: 0
08-10-2011 08:56 PM
Hi ,
This should be an useful link :
https://supportforums.cisco.com/message/3370180#3370180
Thanks,
Priya K
08-10-2011 09:17 PM
Hi Priya,
Thanks for your response. Will changing the 'Send H225 User Info Msg' parameter under H323-Device still apply for a SIP trunk and non-H323 gateway?
Also we currently have several H323 gateways currently in production, so how changing this parameter affect the other gateways?
'Send H225 User Info Msg' is currently set to 'User infor for call progress tone'
Regards,
Ray
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