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raychoiusyd
Beginner

SIP Trunk - no ringback after transfer from PSTN call

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

2 REPLIES 2
priyk
Beginner

Hi ,

This should be an useful link :

https://supportforums.cisco.com/message/3370180#3370180

Thanks,

Priya K

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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