[asterisk-bugs] [Asterisk 0016235]: Placing a URI call fails when URI string contains a non-standard port

[Asterisk Bug Tracker]

The following issue has been RESOLVED. 
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https://issues.asterisk.org/view.php?id=16235 
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Reported By:                test011
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   16235
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     resolved
Asterisk Version:           SVN 
JIRA:                       SWP-419 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 no change required
Fixed in Version:           
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Date Submitted:             2009-11-12 14:22 CST
Last Modified:              2009-12-02 12:10 CST
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Summary:                    Placing a URI call fails when URI string contains a
non-standard port
Description: 
You can make a direct URI call with, say, a softphone register to an
Asterisk like this,

123 at myhome.dyndns.earth

with some entry in extensions.conf like this,

Dial(${EXTEN}@${SIPDOMAIN},.....)

Yes, this works great. Let's assume something else,

123 at myhome.dyndns.earth:8777  (notice the port number at the end)

This does not work. Because ${SIPDOMAIN} does not have ":8777" part of the
URI dialed. It seems Asterisk just throws that part away. If it's not, I
guess it's not documented well enough.

Some document claims that ${SIPDOMAIN} will contain all the string after
'@' But I guess this is not the case.
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 (0114544) file (administrator) - 2009-12-02 12:10
 https://issues.asterisk.org/view.php?id=16235#c114544 
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The SIPDOMAIN channel variable is documented as "SIP destination domain of
an inbound call". It does not say that it contains everything after @ 'nor
does it say that it contains the port.

That being said I would suggest using SIP_HEADER to get the To header and
use that to place your subsequent outgoing call. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-12-02 12:10 file           Note Added: 0114544                          
2009-12-02 12:10 file           Status                   assigned => resolved
2009-12-02 12:10 file           Resolution               open => no change
required
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