[asterisk-bugs] [Asterisk 0012921]: Asterisk 1.4.21 breaks realtime sip on 'sip reload'

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12921 
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Reported By:                Nuitari
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12921
Category:                   PBX/pbx_realtime
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.6.0-beta9 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-06-23 20:59 CDT
Last Modified:              2008-08-15 15:47 CDT
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Summary:                    Asterisk 1.4.21 breaks realtime sip on 'sip reload'
Description: 
Using Asterisk 1.4.21 realtime becomes useless after a sip reload is done.


The dynamic information is cleared, however it doesn't get reloaded from
the database when the friend is doing some activity. The only way to make
the friend show again is to force the phone to register again, usually
though a reboot.

The module is res_mysql, from asterisk-addons 1.4.7, works as expected
with Asterisk 1.4.20.
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---------------------------------------------------------------------- 
 (0091457) bcramer (reporter) - 2008-08-15 15:47
 http://bugs.digium.com/view.php?id=12921#c91457 
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Corydon75, I've applied both patches successfully to the latest 1.4 branch
of chan_sip.c and unfortunately have to report that it still looses the
status information:

*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status    
Realtime  
6046309553/6046309553      24.207.127.215   D   N      5060     OK (58 ms)
Cached RT 
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0
offline]

*CLI> sip reload
Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found

*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status    
Realtime  
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
offline]

If I pickup the phone and dial out I can complete the call:
 -- Executing [6046381111 at fromCLIENT:1] Dial("SIP/6046309553-09240cf0",
"SIP/blah/6046381111") in new stack
    -- Called blah/6046381111
    -- SIP/blah-09248848 answered SIP/6046309553-09240cf0
    -- Packet2Packet bridging SIP/6046309553-09240cf0 and
SIP/blah-09248848
  == Spawn extension (fromCLIENT, 6046381111, 1) exited non-zero on
'SIP/6046309553-09240cf0'

However, the clients status details or registration are still empty.

*CLI>sip show peers
Name/username              Host            Dyn Nat ACL Port     Status    
Realtime  
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
offline]

Note, I'm using ODBC as the datasource, not mysql as others above.  My
sip.conf is extremely small:

[general]
context=default
allowguest=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
nat=yes
rtcachefriends=yes
rtupdate=no
rtautoclear=yes

[authentication] 

Issue History 
Date Modified    Username       Field                    Change               
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2008-08-15 15:47 bcramer        Note Added: 0091457                          
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