[asterisk-bugs] [Asterisk 0017779]: [patch] tcptls.c:350 Unable to connect SIP socket Connection refused

Thu Nov 11 15:58:37 CST 2010

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17779 
====================================================================== 
Reported By:                smallet
Assigned To:                jpeeler
====================================================================== 
Project:                    Asterisk
Issue ID:                   17779
Category:                   Channels/chan_sip/TCP-TLS
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     closed
Target Version:             1.6.2.15
Asterisk Version:           SVN 
JIRA:                       SWP-1999 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             2010-08-02 14:06 CDT
Last Modified:              2010-11-11 15:58 CST
====================================================================== 
Summary:                    [patch] tcptls.c:350 Unable to connect SIP socket
Connection refused
Description: 
Steps to reproduce:

<ol>
<li>Add a peer information as shown in graph 1 below in sip.conf.</li>
<li>do asterisk -r, then sip reload.</li>
<li>That server has connection refused so the asterisk debug will start
complaining it can't connect to the server.</li>
<li>Do sip show peers, you will see the peer with a status of
UNREACHABLE.</li>

So far so good...<br />

<li>Remove The whole graph 1 from your sip.conf file.</li>
<li>Do a sip reload again. the sip show peers will not show the peer
anymore.</li>
<li>The connection refused will keep going on and on forever unless you
actually restart asterisk completely.</li>
</ol>

Is there a way to bypass this or is this a bug?

[Aug  2 14:53:56] ERROR[13926]: tcptls.c:350 ast_tcptls_client_start:
Unable to connect SIP socket to 192.168.0.228:5060: Connection refused
[Aug  2 14:54:10] ERROR[13997]: tcptls.c:350 ast_tcptls_client_start:
Unable to connect SIP socket to 192.168.0.228:5060: Connection refused
[Aug  2 14:54:24] ERROR[14002]: tcptls.c:350 ast_tcptls_client_start:
Unable to connect SIP socket to 192.168.0.228:5060: Connection refused
[Aug  2 14:54:38] ERROR[14011]: tcptls.c:350 ast_tcptls_client_start:
Unable to connect SIP socket to 192.168.0.228:5060: Connection refused
[Aug  2 14:54:52] ERROR[14016]: tcptls.c:350 ast_tcptls_client_start:
Unable to connect SIP socket to 192.168.0.228:5060: Connection refused
[Aug  2 14:55:06] ERROR[14089]: tcptls.c:350 ast_tcptls_client_start:
Unable to connect SIP socket to 192.168.0.228:5060: Connection refused
[Aug  2 14:55:20] ERROR[14099]: tcptls.c:350 ast_tcptls_client_start:
Unable to connect SIP socket to 192.168.0.228:5060: Connection refused
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0016382 [patch] SIP OPTIONS qualify message for...
====================================================================== 

---------------------------------------------------------------------- 
 (0128799) svnbot (reporter) - 2010-11-11 15:58
 https://issues.asterisk.org/view.php?id=17779#c128799 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 294734

_U  branches/1.8/
U   branches/1.8/channels/chan_sip.c

------------------------------------------------------------------------
r294734 | jpeeler | 2010-11-11 15:58:26 -0600 (Thu, 11 Nov 2010) | 32
lines

Merged revisions 294733 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.6.2

................
  r294733 | jpeeler | 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) | 25
lines

  Merged revisions 294688 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4

  ........
    r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) | 18
lines

    Fix problem with qualify option packets for realtime peers never
stopping.

    The option packets not only never stopped, but if a realtime peer was
not in
    the peer list multiple options dialogs could accumulate over time.
This
    scenario has the potential to progress to the point of saturating a
link just
    from options packets. The fix was to ensure that the poke scheduler
checks to
    see if a peer is in the peer list before continuing to poke. The
reason a peer
    must be in the peer list to be able to properly manage an options
dialog is
    because otherwise the call pointer is lost when the peer is
regenerated from
    the database, which is how existing qualify dialogs are detected.

    (closes issue https://issues.asterisk.org/view.php?id=16382)
    (closes issue https://issues.asterisk.org/view.php?id=17779)
    Reported by: lftsy
    Patches: 
          bug16382-3.patch uploaded by jpeeler (license 325)
    Tested by: zerohalo
  ........
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=294734 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-11-11 15:58 svnbot         Checkin                                      
2010-11-11 15:58 svnbot         Note Added: 0128799                          
======================================================================