[asterisk-bugs] [Asterisk 0012921]: Asterisk 1.4.21 breaks realtime sip on 'sip reload'
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Jul 15 07:11:02 CDT 2008
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: bbryant
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Project: Asterisk
Issue ID: 12921
Category: PBX/pbx_realtime
Reproducibility: always
Severity: major
Priority: normal
Status: assigned
Asterisk Version: 1.4.21-rc1
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: 06-23-2008 20:59 CDT
Last Modified: 07-15-2008 07:11 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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Nuitari - 07-15-08 07:10
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*CLI> core show version
Asterisk 1.4.21.1 built by portage @ hammer on a x86_64 running Linux on
2008-07-10 08:52:28 UTC
*CLI> sip reload
*CLI> sip set debug peer 2000
Unable to get IP address of peer '2000' <-- First sign of something
wrong
*CLI> core set debug 99999
Core debug was 8 and is now 99999
*CLI> Really destroying SIP dialog
'144b982e0bb7a39237384a4a27eecf74 at 10.1.8.1' Method: OPTIONS
Really destroying SIP dialog
'71e6a26003c8dd916ca4851560a668f3 at 67.205.71.100' Method: OPTIONS
Really destroying SIP dialog
'681a19556216225f1c9355a735f4b4ee at 67.205.71.100' Method: OPTIONS
-- Executing [didww at incoming-voip:1] Goto("SIP/didww-100761a0",
"mainmenu|s|1") in new stack
--- SNIP DIALPLAN STUFF ---
-- Executing [s at macro-stdexten:300] Dial("SIP/didww-100761a0",
"SIP/2000|25|mw") in new stack
Really destroying SIP dialog
'689592b32dca282b29d5fb90286bd467 at 67.205.71.100' Method: INVITE
[Jul 15 08:10:00] WARNING[8052]: app_dial.c:1183 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s at macro-stdexten:301] Goto("SIP/didww-100761a0",
"s-CHANUNAVAIL|1") in new stack
-- Goto (macro-stdexten,s-CHANUNAVAIL,1)
looking at MySQL's debug log, no query is made to try and load peer 2000.
The problem is clear, Asterisk isn't loading the peer from the database
anymore, as it used to do before 1.4.21.
To replicate it, setup a server to register to the other via SIP.
Use realtime SIP to store the registration.
On the server that is being connected to, do a sip reload.
Try to place a call to the friend in realtime.
Issue History
Date Modified Username Field Change
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07-15-08 07:11 Nuitari Note Added: 0090275
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