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

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Aug 30 15:43:57 CDT 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16112 
====================================================================== 
Reported By:                ajohnson
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16112
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     confirmed
Asterisk Version:           SVN 
JIRA:                       SWP-290 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-10-21 21:14 CDT
Last Modified:              2010-08-30 15:43 CDT
====================================================================== 
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>


====================================================================== 

---------------------------------------------------------------------- 
 (0126447) parisioa (reporter) - 2010-08-30 15:43
 https://issues.asterisk.org/view.php?id=16112#c126447 
---------------------------------------------------------------------- 
I had the same problem because i was missing the mask field in my table
definition.  I set mask to 0.0.0.0/0.0.0.0 and now my peer is able to
register.  

Can somebody explain what exactly mask is supposed to be/do?  I've set
permit/deny like i had in sip.conf with deny=0.0.0.0/0.0.0.0 and permit to
192.168.0.0/255.255.0.0 and that seems to work.  It follows these settings
as well, with mask set to 0.0.0.0.  It appears as though mask doesn't
actually do anything at all. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-30 15:43 parisioa       Note Added: 0126447                          
======================================================================




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