[asterisk-bugs] [Asterisk-GUI 0015142]: sip trunk outboundproxy error.

[Asterisk Bug Tracker]

The following issue has been RESOLVED. 
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https://issues.asterisk.org/view.php?id=15142 
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Reported By:                Romik
Assigned To:                rbrindley
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Project:                    Asterisk-GUI
Issue ID:                   15142
Category:                   Service Providers/Trunks
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     resolved
Asterisk GUI Version:       SVN 
Asterisk Version:           1.4.24 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 4791 
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2009-05-17 18:42 CDT
Last Modified:              2009-05-28 09:37 CDT
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Summary:                    sip trunk outboundproxy error.
Description: 
When adding new SIP trunk in GUI without filling Outbondproxy field in GUI,
it adds the following lines to users.conf:
[trunk_2]
host = host.domain
username = username
secret = secret-pass
trunkname = sipnet  ; GUI metadata
context = DID_trunk_2
group = null
hasexten = no
hasiax = no
hassip = yes
registeriax = no
registersip = yes
trunkstyle = voip
fromdomain = host.domain
fromuser = username
insecure = port,invite
outboundproxy =
disallow = all
allow = gsm,alaw,speex,g726,g723

Blank `outboundproxy =` causes the following errors:
1) trunk_2 does not appear in `sip show peers`
2) and that's why it is not possible to dialout via this trunk:
    -- Executing [1-dial at macro-trunkdial-failover-0.3:1]
Dial("SIP/199-08d16d08", "SIP/trunk_2/89172845917") in new stack
[May 18 03:26:07] WARNING[5531]: chan_sip.c:2984 create_addr: No such
host: trunk_2
[May 18 03:26:07] WARNING[5531]: app_dial.c:1237 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)

If I comment `outboundproxy =` line and do `sip reload` then trunk is
shown in `sip show peers` and it is possible to dial via this trunk.
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---------------------------------------------------------------------- 
 (0105641) svnbot (reporter) - 2009-05-28 09:37
 https://issues.asterisk.org/view.php?id=15142#c105641 
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Repository: asterisk-gui
Revision: 4804

U   branches/2.0/config/js/pbx2.js
U   branches/2.0/config/js/trunks_voip.js

------------------------------------------------------------------------
r4804 | rbrindley | 2009-05-28 09:37:13 -0500 (Thu, 28 May 2009) | 13
lines


- fixed an issue where the GUI would write outboundproxy when it was null
- fixed an issue where the GUI would not delete outboundproxy on edit if
  its new value was null

(closes issue https://issues.asterisk.org/view.php?id=15142)
Reported by: Romik
(closes issue https://issues.asterisk.org/view.php?id=15126)
Reported by: timeshell
(closes issue https://issues.asterisk.org/view.php?id=14986)
Reported by: emrah


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http://svn.digium.com/view/asterisk-gui?view=rev&revision=4804 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-05-28 09:37 svnbot         Note Added: 0105641                          
2009-05-28 09:37 svnbot         Assigned To              awk => rbrindley    
2009-05-28 09:37 svnbot         Status                   assigned => resolved
2009-05-28 09:37 svnbot         Resolution               open => fixed       
======================================================================