[asterisk-bugs] [Asterisk 0016112]: [patch] SIP Realtime not reading database for changes to realtime peers after initial registration

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16112 
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Reported By:                ajohnson
Assigned To:                jpeeler
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Project:                    Asterisk
Issue ID:                   16112
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     assigned
Target Version:             1.6.0.16
Asterisk Version:           1.6.2.0-rc3 
JIRA:                       SWP-290 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-10-21 21:14 CDT
Last Modified:              2009-11-03 17:57 CST
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Summary:                    [patch] SIP Realtime not reading database for
changes to realtime peers after initial registration
Description: 
After a realtime sip peer has registered, changes to the sip_peers table
for the related peer will never be picked up by Asterisk.  In my test I
changed the context from from-test to from-softphone.  The change was not
picked up until after restarting asterisk.

Sip.conf has rtcachefriends=no set.

SELECT * FROM sip_peers WHERE name = 'ajohnson'
5797, 'ajohnson', 'dynamic', 'no', 'friend', '', '', , '', '8448', 'yes',
'yes', 'from-test', '', '', '', '', '', '', '', '', '', '', '', '', '',
'no', '', '', '', '', '97983', '', 'all', 'all', '', '10.210.20.192', 5060,
'pbx2', 1256176828, 'ajohnson', '', 0, 'SJphone/1.65.377a (SJ Labs)'



    -- Registered SIP 'ajohnson' at 10.210.20.192 port 5060
  == Using SIP RTP CoS mark 5
    -- Executing [6027414660 at from-test:1] Dial("SIP/ajohnson-b757d498",
"SIP/sbc/6027414660)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called sbc/6027414660)
  == Spawn extension (from-test, 6027414660, 1) exited non-zero on
'SIP/ajohnson-b757d498'


  == Using SIP RTP CoS mark 5
    -- Executing [6027414660 at from-test:1] Dial("SIP/ajohnson-b7589b98",
"SIP/sbc/6027414660)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called sbc/6027414660)
  == Spawn extension (from-test, 6027414660, 1) exited non-zero on
'SIP/ajohnson-b7589b98'



SELECT * FROM sip_peers WHERE name = 'ajohnson'
5797, 'ajohnson', 'dynamic', 'no', 'friend', '', '', , '', '8448', 'yes',
'yes', 'from-softphone', '', '', '', '', '', '', '', '', '', '', '', '',
'', 'no', '', '', '', '', '97983', '', 'all', 'all', '', '10.210.20.192',
5060, 'pbx2', 1256176828, 'ajohnson', '', 0, 'SJphone/1.65.377a (SJ Labs)'



pbx2*CLI> sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf':   == Found
  == Parsing '/etc/asterisk/sipaccounts.conf':   == Found
  == Parsing '/etc/asterisk/users.conf':   == Found
  == Parsing '/etc/asterisk/sip_notify.conf':   == Found
  == Using SIP RTP CoS mark 5
    -- Executing [6027414660 at from-test:1] Dial("SIP/ajohnson-b7589098",
"SIP/sbc/6027414660)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called sbc/6027414660)
  == Spawn extension (from-test, 6027414660, 1) exited non-zero on
'SIP/ajohnson-b7589098'



pbx2*CLI> sip prune realtime
peer  all
pbx2*CLI> sip prune realtime all
No peers found to prune.
  == Using SIP RTP CoS mark 5
    -- Executing [6027414660 at from-test:1] Dial("SIP/ajohnson-b758da30",
"SIP/sbc/6027414660)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called sbc/6027414660)
  == Spawn extension (from-test, 6027414660, 1) exited non-zero on
'SIP/ajohnson-b758da30'



pbx2*CLI> module reload chan_sip.so
    -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP))
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf':   == Found
  == Parsing '/etc/asterisk/sipaccounts.conf':   == Found
  == Parsing '/etc/asterisk/users.conf':   == Found
  == Parsing '/etc/asterisk/sip_notify.conf':   == Found
  == Using SIP RTP CoS mark 5
    -- Executing [6027414660 at from-test:1] Dial("SIP/ajohnson-b780f5c0",
"SIP/sbc/6027414660)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called sbc/6027414660)
  == Spawn extension (from-test, 6027414660, 1) exited non-zero on
'SIP/ajohnson-b780f5c0'



pbx2*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status    
Realtime
corppbx                    10.210.10.76                5060     OK (1 ms)
inbound/inbound            (Unspecified)    D          5060     UNKNOWN
outbound/outbound          (Unspecified)    D          5060     UNKNOWN
pvpbx                      10.208.8.25                 5060     OK (1 ms)
pvutil                     10.200.10.51                5060     OK (1 ms)
sbc                        10.230.10.90                5060     OK (1 ms)
6 sip peers [Monitored: 4 online, 2 offline Unmonitored: 0 online, 0
offline]
pbx2*CLI>


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---------------------------------------------------------------------- 
 (0113099) jpeeler (administrator) - 2009-11-03 17:57
 https://issues.asterisk.org/view.php?id=16112#c113099 
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I haven't been able to reproduce this either. Does it ever occur on a
single host? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-03 17:57 jpeeler        Note Added: 0113099                          
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