[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.
======================================================================
https://issues.asterisk.org/view.php?id=15142
======================================================================
Reported By: Romik
Assigned To: awk
======================================================================
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:
======================================================================
Date Submitted: 2009-05-17 18:42 CDT
Last Modified: 2009-05-17 18:42 CDT
======================================================================
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.
======================================================================
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
======================================================================
More information about the asterisk-bugs
mailing list