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Summary:ASTERISK-17070: SIP response
Reporter:dogonovmax (dogonovmax)Labels:
Date Opened:2010-12-05 14:12:04.000-0600Date Closed:2011-06-07 14:00:43
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Applications/app_dial
Versions:Frequency of
Occurrence
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Issues:
Environment:Attachments:
Description:Hi!
I have dialplan:

[default]

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

Console:
-- Executing [555@default:3] Dial("SIP/xlite1-00000006", "SIP/004477@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@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 ASTERISK-12437: [patch] Setting up a HANGUPCAUSETEXT variable for SIP channel
but this don't worked for me!!!

Thank you!

Comments:By: Leif Madsen (lmadsen) 2010-12-07 13:30:36.000-0600

That functionality has already been committed to Asterisk, so I would suggest using Asterisk 1.8 to test it. If it doesn't work, please open a new issue with the console output with debug level logging, and the relevant dialplan.

Thanks!