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

Asterisk Bug Tracker noreply at bugs.digium.com
Sun May 17 18:42:15 CDT 2009


The following issue has been SUBMITTED. 
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https://issues.asterisk.org/view.php?id=15142 
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Reported By:                Romik
Assigned To:                awk
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Project:                    Asterisk-GUI
Issue ID:                   15142
Category:                   Service Providers/Trunks
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
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:              
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Date Submitted:             2009-05-17 18:42 CDT
Last Modified:              2009-05-17 18:42 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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Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-05-17 18:42 Romik          New Issue                                    
2009-05-17 18:42 Romik          Status                   new => assigned     
2009-05-17 18:42 Romik          Assigned To               => awk             
2009-05-17 18:42 Romik          Asterisk GUI Version      => SVN             
2009-05-17 18:42 Romik          Asterisk Version          => 1.4.24          
2009-05-17 18:42 Romik          SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
2009-05-17 18:42 Romik          SVN Revision (number only!) => 4791            
======================================================================




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