|Summary:||ASTERISK-06124: SIP PROTOCOL VIOLATION: Route information ignored when sending BYE requests.|
|Reporter:||George Robinson (george robinson)||Labels:|
|Date Opened:||2006-01-18 10:55:04.000-0600||Date Closed:||2006-01-30 04:40:55.000-0600|
|Environment:||Attachments:||( 0) SIP_Debug_log.txt|
|Description:||When Asterisk hangs up an incoming SIP call, the Request-URI of the BYE request is malformed. Thus the hangup() fails.|
This bug happens only when the SIP call comes in through multiple strict-routing SIP proxies.
Apparently, the route information is ignored by Asterisk's SIP implementation, when sending BYE requests.
This bug appears to be related to:
Other SIP implementations (e.g.: OnDO SIP PBX) do not exhibit this bug.
I checked 4 other SIP implementations with the same result.
****** ADDITIONAL INFORMATION ******
Apparently, the route info is properly collected during the original INVITE, as seen below:
list_route: hop: <sip:email@example.com:5066>
list_route: hop: <sip:firstname.lastname@example.org:5060>
list_route: hop: <sip:126.96.36.199:5060>
however the route info above is not used when sending BYE requests.
This is wrong and is the cause of the hangup() error.
The BYE requet that is being transmitted should be:
BYE sip:188.8.131.52:5066 SIP/2.0
|Comments:||By: Olle Johansson (oej) 2006-01-18 11:46:48.000-0600|
Deleted the file.
We have a problem with ALL strict routing that needs to fixed.
By: George Robinson (george robinson) 2006-01-18 11:56:25.000-0600
How long have you been aware of this SIP protocol violation ?
Strict routers are very common in SIP world.
For example the PSTN-SIP gateway provider - Vonage, which has over 1 million subscribers, uses strict SIP proxies.
By: Tilghman Lesher (tilghman) 2006-01-20 10:48:23.000-0600
I'd go so far as to say that this bug is a duplicate of 6240, and we should close this one.
By: Olle Johansson (oej) 2006-01-30 04:40:50.000-0600
Closing this in favour of bug ASTERISK-6082 that also covers strict routing.