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Non tromboned SIP transfer

I have a third party app connected via a SIP trunk to me CUCM 8.6.2 system.

The app receives a call, makes a second call back to CUCM, then issues a SIP REFER to try to get CUCM to transfer the call, so the app can get out of the way.

However, the app gets back a "404 Not Found" from CUCM.

Any suggestions as to how to troubleshoot this ? I've looked at the SIP traces on CUCM, and I can't see anything jumping out at me.

GTG

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Non tromboned SIP transfer

Hi Gordon

Can you look at your SIP trunk on CUCM

Have you set a CSS in this field

Rerouting Calling Search Space

Calling search spaces determine the partitions that calling devices can search when they attempt to complete a call. The rerouting calling search space gets used to determine where a SIP user (A) can refer another user (B) to a third party (C). After the refer is completed, B and C connect. In this case, the rerouting calling search space that is used is that of the initial SIP user (A).

Note Calling Search Space also applies to 3xx redirection and INVITE with Replaces features.

The default value for Rerouting Calling Search Space specifies None.


May help.

Regards

Alex

Regards, Alex. Please rate useful posts.
2 REPLIES
Green

Non tromboned SIP transfer

Hi Gordon

Can you look at your SIP trunk on CUCM

Have you set a CSS in this field

Rerouting Calling Search Space

Calling search spaces determine the partitions that calling devices can search when they attempt to complete a call. The rerouting calling search space gets used to determine where a SIP user (A) can refer another user (B) to a third party (C). After the refer is completed, B and C connect. In this case, the rerouting calling search space that is used is that of the initial SIP user (A).

Note Calling Search Space also applies to 3xx redirection and INVITE with Replaces features.

The default value for Rerouting Calling Search Space specifies None.


May help.

Regards

Alex

Regards, Alex. Please rate useful posts.

Re: Non tromboned SIP transfer

Thanks for that. Works like a charm.

Now for the follow-up questions

1 - Why does a SIP trunk need a reroute calling search space for transfers to work, but a phone (which also has the option for a reroute CSS) doesn't ?

2 - Why would you want different CSSs for normal inbound calls, and for transfered calls ?

3 - What's the difference between the Rerouting CSS & the "Out-of-dialog Refer CSS" ?

GTG

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