CUBE - unusual SIP cause code from ITSP in the BYE request

[mcaldogne]

Hello!

We have some problems by a customers with CISCO2921/K9 as a CUBE for SIP to SIP calls.

When the BYE comes from the external user on ITSP side, in the SIP message is the following the reason code:

Reason: Q.850;cause=16;text="Normal call clearing",SIP;cause=699;text="Unknown"

The second "piece" of the reason code  - ",SIP;cause=699;text="Unknown" - cause problem on customers side, because the customer's server  recognized this as an error.

The ITSP sad us they apparently cannot change this setting in the BYE message, though I think this 699 does not exist as a standard SIP message (6XX should be clobal failure).

Here the whole SIP message coming from the ITSP:

.Nov 3 11:38:37.887: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:047******2@172.19.237.251:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.53.69:5060;branch=z9hG4bKb076j9vgv15o1jjrvui5tbe0u0-g1k5
CSeq: 3 BYE
From: <sip:047******1@10.247.10.40;user=phone>;tag=4c58da5b
To: <sip:047******2@10.247.15.153;lr>;tag=8E2E2414-2468
Call-ID: 00c6380700c637f-0002-040a-0000-0000@10.2.9.203
P-Charging-Vector: icid-value="00c6384c00c6383-0002-040a-0000-0000@10.2.9.203";icid-generated-at=10.247.10.40;orig-ioi=10.247.10.40
Supported: 100rel
Max-Forwards: 68
Contact: <sip:047******1@192.168.53.69:5060;transport=udp>
Reason: Q.850;cause=16;text="Normal call clearing",SIP;cause=699;text="Unknown"
Content-Length: 0

It would be possibile to change the reason code on the CUBE to cut this strange SIP cause as the call will be routed over the internal dial-peer to the customer server?

Thanks in advanced for your help.

Mirko

[b.winter · Accepted solution]

Hi,

you could use a SIP-profile on the CUBE, to strip it.

Something like:

voice class sip-profiles <nr>

 rule 1 request BYE sip-header Reason modify "(.*),SIP;.*" "\1;"

But I'm currently not sure, if you need the ending semicolon or not. Probably you have to check on that. I think, it isn't needed.

[mcaldogne]

Hello,

we fixed the problem by creating a new voice class SIP-profile for the dial-peer.

Thank you very much for your help.

Mirko

[b.winter]

Hello Mirko,

that sounds great. No problem.

I would be glad, if you could mark the post as a solution

BR, Björn

[mcaldogne]

OK, no problem!

Here also the final configuration of the rule:

voice class sip-profiles 4
request BYE sip-header Reason modify ",SIP;.*" ""