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

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Mar 9 12:30:36 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 12:30 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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---------------------------------------------------------------------- 
 (0101401) file (administrator) - 2009-03-09 12:30
 http://bugs.digium.com/view.php?id=14626#c101401 
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Okay, if it was totally broken then all addresses would fail to use the
port. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-09 12:30 file           Note Added: 0101401                          
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