Cisco unity connection over sip transfer failure

[Austin3:16]

Hello, i have set up a sip trunk cucm to unity connection, the css of the sip trunk and vm pilot is INTERNAL_CSS, this css contain INTERNAL_PATRITION, aftr that i have configure a Auto Attendant and after i call the auto attendant from100  and dial a internal number 101 they told me that this extension is not available only if i add the internal partition on the line of 101 however if i change the partition to none the call transfer wthout any problem, 

any idea for that ? need your help 

[MrButton]

what's happening bro.  Can you tell us the call flow, PT, any cti route points, and CSS just for what it needs to do?  Also run a route plan report for this AA (transfer rules).  sounds like you have everything in the same PT/CSS.  how unity is handling the call. 101 needs to be be reachable.  Is it tied to a device?  can you call 101 from any other device with a DN?  If the none is selected this means the other DNs/CTI or whatever number those are is on some other partition.  You can also run DNA: https://yourcucm/dna/ to look at the call flow.

[Jonathan Schulenberg · Accepted solution]

The CUCM SIP Trunk is likely missing the Rerouting CSS. Look further down the page and ensure it is set in addition to the Incoming CSS. Transfers trigger a SIP REFER which is subject to the Rerouting CSS.

[Austin3:16]

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@Jonathan Schulenberg wrote:

The CUCM SIP Trunk is likely missing the Rerouting CSS. Look further down the page and ensure it is set in addition to the Incoming CSS. Transfers trigger a SIP REFER which is subject to the Rerouting CSS.

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@Jonathan Schulenberg wrote:

The CUCM SIP Trunk is likely missing the Rerouting CSS. Look further down the page and ensure it is set in addition to the Incoming CSS. Transfers trigger a SIP REFER which is subject to the Rerouting CSS.

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Thank you its work