[asterisk-bugs] [Asterisk 0013121]: Sip to Sip dial and rtsavesysname not working in latestsvn

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
====================================================================== 
http://bugs.digium.com/view.php?id=13121 
====================================================================== 
Reported By:                kactus
Assigned To:                mnicholson
====================================================================== 
Project:                    Asterisk
Issue ID:                   13121
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 132312 
Request Review:              
Resolution:                 fixed
Fixed in Version:           1.6.1
====================================================================== 
Date Submitted:             2008-07-21 08:08 CDT
Last Modified:              2009-01-28 17:03 CST
====================================================================== 
Summary:                    Sip to Sip dial and rtsavesysname not working in
latestsvn
Description: 
Hello everyone

Have been playing with SVN-trunk-r132312M, looks like it has fixed alot of
the issues I experienced before, but I have come across two problems.

Firstly dialing a realtime sip (no rtcache) such as Account1 dialing
Account2 produces the output:


    -- Executing Dial("SIP/Account1-0880b814", "SIP/Account2")
[Jul 21 20:33:08] WARNING[46712]: app_dial.c:1476 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Dial("SIP/Account1-0880b814", "SIP/Account2")
[Jul 21 20:33:08] WARNING[46712]: app_dial.c:1476 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Dial("SIP/Account1-0880b814", "SIP/Account2")
[Jul 21 20:33:08] WARNING[46712]: app_dial.c:1476 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/Account1-0880b814' status is
'CHANUNAVAIL'
    -- Executing Hangup("SIP/Account1-0880b814", "")

The same box, with the same configs running 1.4.17 produces this:

    -- Executing Dial("SIP/Account1-3", "SIP/Account2")
    -- Called Account2
    -- SIP/Account2-087b0000 is ringing

Quite a bit of a problem especially when inbound call termination occurs
on another box and is dialed at your box as an extension. I experienced the
exact same behaviour when an IAX peer was dialing a sip extension (ie exten
=> NUM,1,Dial(IAX2/Account:Pass at box2/accounttodial)) it worked in previous
betas of 1.6 as well.

Secondly sip peers that registered successfully in the SVN did not update
the regserver field in the sip table while they did under 1.4.17 (and betas
of 1.6)

Please let me know if you need any more info.

All the best - Kamil


======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0014205 Unable to register SIP device with real...
====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-28 17:03 blitzrage      Resolution               open => fixed       
2009-01-28 17:03 blitzrage      Fixed in Version          => 1.6.1           
======================================================================