[asterisk-bugs] [Asterisk 0013638]: SIP with canreinvite=yes through multiple Asterisk

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13638 
====================================================================== 
Reported By:                mpiazzatnetbug
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13638
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.22 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-10-07 09:48 CDT
Last Modified:              2008-10-07 12:21 CDT
====================================================================== 
Summary:                    SIP with canreinvite=yes through multiple Asterisk
Description: 
This issue is similar to the BUG number 0010481. I have two asterisk
identical each other, and a Openser in the middle to route the call based
on prefix (in product will be noumerous asterisk and some other sip PBX).
Every device is in the same network, so no NAT issue. 
Thia is the scenario

phone1 --> Asterisk1 --> Openser --> Asterisk2 --> phone2
The asterisk is registred on Openser with a Sip trunk, this is the sip
conf for the asterisk:


[phone1]
type=friend
username=phone1
callerid=("" <2678>)
secret=1234
context=users-distretto
callgroup=1
pickupgroup=1
canreinvite=yes
nat=no
host=dynamic


[siptrunk]
type=friend
username=8889
secret=8889
fromuser=8889
host=siptrunk.tnet.it
dtmfmode=rfc2833
fromdomain=siptrunk.tnet.it
context=default
insecure=very
canreinvite=yes

and on the dialplan to dial
exten => _04.,n,Dial(SIP/${EXTEN}@siptrunk)
and to recieve
exten => _04.,1,Goto(user|${EXTEN}|1)

If I have enable canreinvite=yes in the general section, on the phone and
on the siptrunk I can see a lot of 491 message and the RTP stream in not
forwarded correctly.

Of course If I choose canreinvit=no every were evithing is fine.

If I select canreinvite=no in general , and in the user and on the
siptrunk =yes the phone 1 send the rtp stream directly to the asterisk2 and
the phone 2 send the RTP stream to asterisk2.




 
====================================================================== 

---------------------------------------------------------------------- 
 (0093293) blitzrage (administrator) - 2008-10-07 12:21
 http://bugs.digium.com/view.php?id=13638#c93293 
---------------------------------------------------------------------- 
You will need to provide SIP traces and SIP history in order to move this
bug along. Also please provide a very brief dump of the 'rtp debug' from
the console. Please also provide the relevant portions of the OpenSER
configuration in order to determine if this is a configuration issue or
bug.

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-10-07 12:21 blitzrage      Note Added: 0093293                          
======================================================================