[asterisk-bugs] [Asterisk 0018366]: [patch] SIP/TCP phones are not added to astdb - causes sip reload problems

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18366 
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Reported By:                MKemner
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18366
Category:                   Channels/chan_sip/TCP-TLS
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.8.1-rc1 
JIRA:                       SWP-2648 
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:             2010-11-24 05:46 CST
Last Modified:              2010-12-16 17:10 CST
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Summary:                    [patch] SIP/TCP phones are not added to astdb -
causes sip reload problems
Description: 
Phones that are registered via TCP are not added to the
/SIP/Registry database.  As a result, these phones "vanish" (become
unregistered) after a sip reload and can not be called by asterisk until
they re-register.

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 (0129710) vois (reporter) - 2010-12-16 17:10
 https://issues.asterisk.org/view.php?id=18366#c129710 
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i read the notes made following changes in chan_sip.c. 

Original code
if (!peer->rt_fromcontact && (peer->socket.type & SIP_TRANSPORT_UDP))

New code
if (!peer->rt_fromcontact && (SIP_TRANSPORT_UDP))

and i am running fine from last 1 hours. I have almost 600(400 TLS and 200
UDP) peers in this server. Did sip reload many times. If i find any bug i
will update. Also i noticed big changed in CPU consumption. Without this
asterisk was almost @ 40-50% of CPU consumption and now asterisk is @
10-20% of CPU usage. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-12-16 17:10 vois           Note Added: 0129710                          
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