[asterisk-bugs] [Asterisk 0018428]: SIP response

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18428 
====================================================================== 
Reported By:                dogonovmax
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18428
Category:                   Applications/app_dial
Reproducibility:            have not tried
Severity:                   feature
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.2.14 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-12-05 14:12 CST
Last Modified:              2010-12-05 14:12 CST
====================================================================== 
Summary:                    SIP response
Description: 
Hi! 
I have dialplan: 

[default]

exten => 555,n,Answer
exten => 555,n,Dial(SIP/004477 at sipnet.ru,,tTwWg)

Console: 
 -- Executing [555 at default:3] Dial("SIP/xlite1-00000006",
"SIP/004477 at sipnet.ru,,tTwWg") in new stack
  == Using SIP RTP CoS mark 5
       > ast_get_srv: SRV lookup for '_sip._udp.sipnet.ru' mapped to host
sipnet.ru, port 5060
    -- Called 004477 at sipnet.ru
    -- Got SIP response 480 "No address found" back from 212.53.40.40
    -- SIP/sipnet.ru-00000007 is circuit-busy


And I need receive SIP response "Got SIP response 480 "No address found"
back from 212.53.40.40" in variable after Dial command. How I can take that
in variable? Thank you!

I try patch https://issues.asterisk.org/view.php?id=13140: [patch] Setting up a
HANGUPCAUSETEXT variable for SIP
channel
but this don't worked for me!!!

Thank you!

 
====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-12-05 14:12 dogonovmax     New Issue                                    
2010-12-05 14:12 dogonovmax     Asterisk Version          => 1.6.2.14        
2010-12-05 14:12 dogonovmax     Regression                => No              
2010-12-05 14:12 dogonovmax     SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
======================================================================