SIP INVITE Loop over SIP Trunk

[marco fina]

Hi everybody,

i have a sip trunk between cisco callmanager 9 and asterisk. we want to move extensions from asterisk to cisco callmanager.

For migration purposes we are using this approach: if an extension is not found on cisco callmanager then sip trunk toward asterisk is invoked, and

if an extension is not found on asterisk then sip trunk toward cisco callmanager is invoked. Now the problem is when an inexistent extension is dialed (check this behaviour in packet capture attached).

I know i can prevent loop by configuring css and partition, but i want to know if there is a way to prevent cisco callmanager from searching the same

extension repeatedly in the captured scenario .

i know is a strange request but curious about this

regards

marco

[marco fina]

172.23.112.20 is cisco callmanager

172.23.112.10 is asterisk

[Chris Deren · Accepted solution]

Agree with Jameson here, use CSS on the trunk that would prevent tramboning would be the easiest solution. You could look into implementing QSIG over SIP which is supported on newer versions of CUCM (if I recall 8.6 or 9+) if Asterisk can do it, as QSIG can avoid loops natively.

[marco fina]

Thanks all,

css/partition it's the best solution.

regards

marco

[Chris Deren]

Great, let us know how it works out by rating/marking the posts.

Chris

[marco fina]

Sure,

with css\partition approach now i see only 3 INVITE when an unknown number is dialed, in the previous capture 100 INVITE (loop scenario).

a brief description of a simple environment:

Route Pattern 591XXX that point to SIP Trunk toward Asterisk site (let's say SIP_Trunk-Asterisk). i moved this RP from partition internal let's say Pt_Internal to another partition let's say Pt_X (this moving solves loop).

All extension migrated from Asterisk to Cisco callmanager are in partition Pt_Internal.

All devices registered to cisco callmanager can point to this RP except for SIP_Trunk-Asterisk that has an inbound css that can open Pt_Internal only.

Let's go for an example:

From asterisk an unknown number is dialed, this number is carried to the cisco callmanager through incoming SIP_Trunk-Asterisk, because of css applied on trunk RP 591XXX  can not be selected --> so no loop

From cisco phone an unknown number is dialed, RP 591XXX is selected and the call is carried to the asterisk , asterisk does not know the number so it come back to cisco. Search stops here.

per unknown i mean no existent.

regards

marco

[Jameson Gagnepain · Accepted solution]

Marco,

The simplest solution here would be to ensure that the CSS used by the SIP trunk between Asterisk and CUCM does not include the partition which the SIP trunk is in.

From your capture, it looks like Asterisk drops the Cisco call identifiers when sending the call back... so Cisco wouldn't have a good way to recognize that it's the same call. A Partition/CSS change really seems like the best call here.

-Jameson