[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 16:44 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.
====================================================================== 

---------------------------------------------------------------------- 
 (0091460) bcramer (reporter) - 2008-08-15 16:44
 http://bugs.digium.com/view.php?id=12921#c91460 
---------------------------------------------------------------------- 
Corydon76, Outbound calls work, but not inbound as the client is not in a
reachable state:


Inbound Call 1:
    -- Registered SIP '6046309553' at 24.207.127.0 port 5060 expires 3600
[Aug 15 07:47:01] NOTICE[30791]: chan_sip.c:12703
handle_response_peerpoke: Peer '6046309553' is now Reachable. (68ms /
2000ms)
    -- Executing Macro("SIP/blah-08a86990",
"dialAcctInternal|SIP/6046309553|500|bcramer")
    -- Executing [s at macro-dialAcctInternal:1] GotoIf("SIP/blah-08a86990",
"0?notinservice|s|1") in new stack
    -- Executing [s at macro-dialAcctInternal:2] Set("SIP/blah-08a86990",
"CDR(accountcode)=bcramer") in new stack
    -- Executing [s at macro-dialAcctInternal:3] Dial("SIP/blah-08a86990",
"SIP/6046309553|500|r") in new stack
    -- Called 6046309553
    -- SIP/6046309553-08a8bda8 is ringing
  == Spawn extension (macro-dialAcctInternal, s, 3) exited non-zero on
'SIP/blah-08a86990' in macro 'dialAcctInternal'
  == Spawn extension (macro-dialAcctInternal, s, 3) exited non-zero on
'SIP/blah-08a86990'
    -- Executing [h at macro-dialAcctInternal:1] Hangup("SIP/blah-08a86990",
"") in new stack
  == Spawn extension (macro-dialAcctInternal, h, 1) exited non-zero on
'SIP/blah-08a86990'

sbc02*CLI> sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found

sbc02*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status    
Realtime  
blah/blah  10.10.200.10     D   N      5060     OK (1 ms)            
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0
offline]


Inbound Call 2:
    -- Executing Macro("SIP/blah-b7a04b28",
"dialAcctInternal|SIP/6046309553|500|bcramer")
    -- Executing [s at macro-dialAcctInternal:1] GotoIf("SIP/blah-b7a04b28",
"0?notinservice|s|1") in new stack
    -- Executing [s at macro-dialAcctInternal:2] Set("SIP/blah-b7a04b28",
"CDR(accountcode)=bcramer") in new stack
    -- Executing [s at macro-dialAcctInternal:3] Dial("SIP/blah-b7a04b28",
"SIP/6046309553|500|r") in new stack
[Aug 15 07:48:06] WARNING[30878]: app_dial.c:1183 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 [s at macro-dialAcctInternal:4] Goto("SIP/blah-b7a04b28",
"s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-dialAcctInternal,s-CHANUNAVAIL,1)
  == Auto fallthrough, channel 'SIP/blah-b7a04b28' status is
'CHANUNAVAIL'

If I pick up the device SIP/6046309553 and call out from it, it works as
advertised.  During the sip reload asterisk destroy's all the registrations
then reloads the sip.conf with new information.  It should only do that for
the static clients, the dynamic ones when the caching in enabled should be
updated from the DB with the only exception being the last IP and latency
info, which I believe is still being saved. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-08-15 16:44 bcramer        Note Added: 0091460                          
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