[Asterisk-Users] SIP connections do not hang up

[Ian Hailey]

Hello all,

I also have this SIP CANCEL problem and have inverstigated the problem a 
bit but am not sure if the problem lies in the sipgate proxy or asterisk:

1.) This only happens when you CANCEL an INVITE (obviously) the INVITE 
is shown below.
2.) sipgate sends a 183 Session Prgress response message to asterisk 
shown below as an example.
3.) To hand up Asterisk sends a CANCEL message to sipgate and it is 
happy and acknowledges.
4.) The called parties phone contiunes to ring (incorrectly)!
5.) The problem happens because the CANCEL SIP line contains a different 
contact to the original INVITE massage which according to the SIP RFC 
3261 is illegal.

from RFC 3261  chapter 9.1

"The Request-URI, Call-ID, To, the numeric part of CSeq, and From header
fields in the CANCEL request MUST be identical to those in the
request being cancelled, including tags."

I assume the Request-URI in this case is defined as 
sip:07315094436 at sipgate.de in my exaple INVITE message. The CANCEL 
contains the URI which can be found in the 183 Session Progress message 
set to be sip:497315094436 at 217.10.79.30.

Asterisk chan_sip has the following code part that handles all other 
SIP/2.0 messages (e.g. not specifically processed) in function 
handle_request():

    } else if (!strcasecmp(cmd, "SIP/2.0")) {
        extract_uri(p, req);

The function extract_uri updates the URI of the current context, as this 
has changed in the 183 Session Progress then the URI used in the CANCEL 
message also changes.

My conclusion is the Asterisk probably handles this wrongly as it does 
not happen with soft phones (they use the original INVITE URI). I have 
curently commented out the code part and it fixes the problem I have 
described BUT I do not know enough about the code to say why it was 
included in the first place! So it would be nice to submit this problem 
to the developer to get his answer and let us know what he/she thinks 
should happen. Does anyone know how to submit this kind of request?

Let me know your opinions.

Ian Hailey.

INVITE sip:07315094436 at sipgate.de SIP/2.0
Via: SIP/2.0/UDP 217.187.52.208:5060;branch=z9hG4bK748c9b3f
From: "5337478" <sip:5337478 at 217.187.52.208>;tag=as31d0bbe0
To: <sip:07315094436 at sipgate.de>
Contact: <sip:5337478 at 217.187.52.208>
Call-ID: 56bb3e225ed45640719fe08c244deb0a at 217.187.52.208
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 13 Aug 2004 05:50:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 243

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 217.187.52.208:5060;branch=z9hG4bK748c9b3f
From: "5337478" <sip:5337478 at 217.187.52.208>;tag=as31d0bbe0
To: <sip:07315094436 at sipgate.de>;tag=as0deea40d
Call-ID: 56bb3e225ed45640719fe08c244deb0a at 217.187.52.208
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:497315094436 at 217.10.79.30>
Content-Type: application/sdp
Content-Length: 240

CANCEL sip:497315094436 at 217.10.79.30 SIP/2.0
Via: SIP/2.0/UDP 217.187.52.208:5060;branch=z9hG4bK748c9b3f
From: "5337478" <sip:5337478 at 217.187.52.208>;tag=as31d0bbe0
To: <sip:07315094436 at sipgate.de>
Contact: <sip:5337478 at 217.187.52.208>
Call-ID: 56bb3e225ed45640719fe08c244deb0a at 217.187.52.208
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


Florian Rau wrote:

>Hi,
>
>Well,  the Problem is not the ZAP Channel but the SIP Channel, because it
>occurs no matter what channel I use the phone outside. Maybe this graph is
>more descriptive:
>
>1. ZAP or SIP ==> 2. Asterisk ==> 3. SIP (thru internet, provider sipgate)
>==> 4. PSTN
>
>The connections on 1. hang up correctly, as seen in the log, but the SIP
>connection of 3. does NOT hangup.
>
>Regards,
>Florian
>
>PS: Believe me, I'm searching for over one week in the whole internet for a
>solution, but did not find it.
>
>
>
>----- Original Message ----- 
>From: "Jean-Yves Avenard" <jean-yves.avenard at hydrix.com>
>To: <asterisk-users at lists.digium.com>
>Sent: Friday, July 30, 2004 11:46 PM
>Subject: Re: [Asterisk-Users] SIP connections do not hang up
>
>
>  
>
>>-----BEGIN PGP SIGNED MESSAGE-----
>>Hash: SHA1
>>
>>If you just bothered to search this list in the past 12 hours, you
>>would have found a solution around that:
>>
>>to summarize:
>>
>>Add in zapata.conf:
>>busydetect=yes
>>busycount=6
>>
>>The maximum it will take for asterisk to see the person hanged-up is
>>after 6 busy dial-tones.
>>
>>On 31/07/2004, at 6:58 AM, Florian Rau wrote:
>>
>>    
>>
>>>I'm calling from inside (either X-Lite using SIP channel or a ISDN
>>>telephone
>>>using Zap Channel) using sipgate to a number in public network.
>>>When I'm hanging up before the other person picked up the phone, the
>>>line is
>>>not closed correctly.
>>>The phone keeps on ringing until timeout (of Sipgate I assume) and it
>>>even
>>>costs my money, if the other person picks up the ringing phone, even
>>>if I
>>>already hung up.
>>>
>>>      
>>>
>>- ---
>>Jean-Yves Avenard
>>Hydrix Pty Ltd - Embedding the net
>>www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724
>>-----BEGIN PGP SIGNATURE-----
>>Version: GnuPG v1.2.4 (Darwin)
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>>uuKZxyJoaSmpI9V9I+ojnJc=
>>=Y8jQ
>>-----END PGP SIGNATURE-----
>>
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>
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