[asterisk-bugs] [Asterisk 0014626]: SIP dial ignores destination port
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Mar 9 08:57:52 CDT 2009
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=14626
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Reported By: acunningham
Assigned To:
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Project: Asterisk
Issue ID: 14626
Category: Channels/chan_sip/General
Reproducibility: sometimes
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.6.0.6
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-03-08 21:05 CDT
Last Modified: 2009-03-09 08:57 CDT
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Summary: SIP dial ignores destination port
Description:
When sending this from an AGI to Asterisk 1.6.0.6:
EXEC Dial SIP/phone_123456 at 1.2.3.4:5070,3600,ot
The Asterisk console reports:
-- AGI Script Executing Application: (Dial) Options:
(SIP/phone_123456 at 1.2.3.4:5070,3600,ot)
but then sends to 5060:
Reliably Transmitting (NAT) to 1.2.3.4:5060:
INVITE sip:phone_123456 at 1.2.3.4 SIP/2.0
Do I have the wrong format for the Dial?
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(0101372) file (administrator) - 2009-03-09 08:57
http://bugs.digium.com/view.php?id=14626#c101372
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Have you tried simplifying things and tried it outside the scope of all of
that with another IP address/port?
Issue History
Date Modified Username Field Change
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2009-03-09 08:57 file Note Added: 0101372
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