[asterisk-bugs] [Asterisk 0014626]: SIP dial ignores destination port

[Asterisk Bug Tracker]

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 19:43 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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---------------------------------------------------------------------- 
 (0101441) acunningham (reporter) - 2009-03-09 19:43
 http://bugs.digium.com/view.php?id=14626#c101441 
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After doing some more testing, I'm pretty sure that this is the same
problem as http://bugs.digium.com/view.php?id=13355, but for Asterisk 1.6 rather
than 1.4. Calling any other
IP address than the one in sip.conf works correctly.

Would it be possible to have the patch in
http://bugs.digium.com/view.php?id=13355 ported to 1.6 and
applied? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-09 19:43 acunningham    Note Added: 0101441                          
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