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

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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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 08:47 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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 (0101369) file (administrator) - 2009-03-09 08:47
 http://bugs.digium.com/view.php?id=14626#c101369 
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I've just confirmed under both 1.6.0.6 and 1.6.0 branch that the Dial line
specified with port works as expected. Is it perhaps configured in
sip.conf? Can you provide the complete console output? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-09 08:47 file           Note Added: 0101369                          
2009-03-09 08:47 file           Status                   new => feedback     
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