[asterisk-bugs] [Asterisk 0018437]: "transmit_refer" builds REFER with no host in case of anonymous call transfer

[Asterisk Bug Tracker]

The following issue has been CLOSED 
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https://issues.asterisk.org/view.php?id=18437 
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Reported By:                kpect
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18437
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             2010-12-08 09:33 CST
Last Modified:              2010-12-16 11:19 CST
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Summary:                    "transmit_refer" builds REFER with no host in case
of anonymous call transfer
Description: 
In case of Anonymous call Invite comes with the following "From" field:

> From: Anonymous <sip:192.168.0.1>

Transfer is called without specifying host.

> exten => 1.,1,Transfer(100)


Refer is build in function "transmit_refer" (line 12683 of chan_sip
module)
----
/* Get just the username part */
if ((c = strchr(dest, '@'))) {
	c = NULL;
} else if ((c = strchr(of, '@'))) {
	*c++ = '\0';
}
if (c) {
	snprintf(referto, sizeof(referto), "<sip%s:%s@%s>", use_tls ? "s" : "",
dest, c);
} else {
	snprintf(referto, sizeof(referto), "<sip%s:%s>", use_tls ? "s" : "",
dest);
}
----

In mentioned case "dest" is "100" and "of" is "192.168.0.1".

So Refer-To filed contains "Refer-To: <sip:100>" without specifying host
address.
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---------------------------------------------------------------------- 
 (0129691) lmadsen (administrator) - 2010-12-16 11:19
 https://issues.asterisk.org/view.php?id=18437#c129691 
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This looks like a configuration issue. The destination should be an IP
address, not an extension number. It may work is [100] is defined in
sip.conf, but you would probably need to specify the SIP/100.

Please use the asterisk-users mailing list or #asterisk IRC channel for
additional support. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-12-16 11:19 lmadsen        Note Added: 0129691                          
2010-12-16 11:19 lmadsen        Status                   new => closed       
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