[Asterisk-Users] SIP debugging

[Andres]

Hi Klaus,

The response to a CANCEL should be a "487 Request Terminated", not  a 
"200 OK".  Maybe your innovaphone Server is to blame.

-- 
Andres
Technical Support
http://www.telesip.net




Klaus Darilion wrote:

> Hi!
>
> I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does 
> not accept the 200 OK responses. E.g in the following example, 
> Asterisk retransmits the CANCEL although the 200 OK is received.
>
> There is no log message, why this packet is not accepted/processed. Is 
> there a ways to increase the sip debugging?
>
> thanks
> klaus
>
> Retransmitting #5 (NAT) to 192.174.68.4:5060:
> CANCEL sip:431505641636 at 192.174.68.4 SIP/2.0
> Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport
> From: "klaus1" <sip:00437248600777 at 213.174.230.213>;tag=as4233f839
> To: <sip:431505641636 at 192.174.68.4>
> Contact: <sip:00437248600777 at 213.174.230.213>
> Call-ID: 4c614a3a4a2d5ba94ea4be5e62c3d37f at 213.174.230.213
> CSeq: 102 CANCEL
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
> poeast01*CLI>
> <-- SIP read from 192.174.68.4:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport
> From: "klaus1" <sip:00437248600777 at 213.174.230.213>;tag=as4233f839
> To: <sip:431505641636 at 192.174.68.4>;tag=2870350146
> Call-ID: 4c614a3a4a2d5ba94ea4be5e62c3d37f at 213.174.230.213
> CSeq: 102 CANCEL
> Server: innovaphone IP800 / V6.00 dvl [06-60123]
>
> --- (7 headers 0 lines)---
> Destroying call '4c614a3a4a2d5ba94ea4be5e62c3d37f at 213.174.230.213'
> Retransmitting #6 (NAT) to 192.174.68.4:5060:
> CANCEL sip:431505641636 at 192.174.68.4 SIP/2.0
> Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport
> From: "klaus1" <sip:00437248600777 at 213.174.230.213>;tag=as4233f839
> To: <sip:431505641636 at 192.174.68.4>
> Contact: <sip:00437248600777 at 213.174.230.213>
> Call-ID: 4c614a3a4a2d5ba94ea4be5e62c3d37f at 213.174.230.213
> CSeq: 102 CANCEL
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
> poeast01*CLI>
> <-- SIP read from 192.174.68.4:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK6a092547;rport
> From: "klaus1" <sip:00437248600777 at 213.174.230.213>;tag=as4233f839
> To: <sip:431505641636 at 192.174.68.4>;tag=2870350146
> Call-ID: 4c614a3a4a2d5ba94ea4be5e62c3d37f at 213.174.230.213
> CSeq: 102 CANCEL
> Server: innovaphone IP800 / V6.00 dvl [06-60123]
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>