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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14626 
====================================================================== 
Reported By:                acunningham
Assigned To:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2009-03-08 21:05 CDT
Last Modified:              2009-03-10 08:33 CDT
====================================================================== 
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?
====================================================================== 

---------------------------------------------------------------------- 
 (0101462) svnbot (reporter) - 2009-03-10 08:33
 http://bugs.digium.com/view.php?id=14626#c101462 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 180799

U   branches/1.6.0/channels/chan_sip.c

------------------------------------------------------------------------
r180799 | file | 2009-03-10 08:32:59 -0500 (Tue, 10 Mar 2009) | 5 lines

If a port is specified when dialing a peer then use it.

(closes issue http://bugs.digium.com/view.php?id=14626)
Reported by: acunningham

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=180799 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-10 08:33 svnbot         Note Added: 0101462                          
======================================================================