[asterisk-bugs] [Asterisk 0017305]: language resets during blind transfer

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17305 
====================================================================== 
Reported By:                jamicque
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   17305
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.1.19 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-05-07 03:53 CDT
Last Modified:              2010-05-07 03:53 CDT
====================================================================== 
Summary:                    language resets during blind transfer
Description: 
I've noticed that when B side of the call makes a blindxfer the language
played in phone is always in English despite of the set variables on the
channels.

Here is my example dialplan:

exten => h,2,Hangup
exten => t,1,Hangup
exten => T,1,Hangup
exten => _.,1,Set(CHANNEL(language)=pl)
exten => _.,2,Set(LANGUAGE()=pl)
exten => 11,3,Dial(SIP/test003,20,Tt)
exten => 12,3,Dial(SIP/test004,20,Tt)
exten => 13,3,Dial(SIP/test005,20,Tt)
exten => _.,n,Hangup


and here are the logs:
    -- Executing [11 at CALLEX:1] Set("SIP/test005-00000021",
"CHANNEL(language)=pl") in new stack
    -- Executing [11 at CALLEX:2] Set("SIP/test005-00000021",
"LANGUAGE()=pl") in new stack

    -- Executing [11 at CALLEX:3] Dial("SIP/test005-00000021",
"SIP/test003,20,Tt") in new stack
  == Using SIP RTP TOS bits 136
  == Using SIP RTP CoS mark 4
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 4
  == Using UDPTL TOS bits 136
  == Using UDPTL CoS mark 4
    -- Called test003
    -- SIP/test003-00000022 is ringing
    -- SIP/test003-00000022 answered SIP/test005-00000021
    -- Started music on hold, class 'default', on SIP/test005-00000021
    -- <SIP/test003-00000022> Playing 'pbx-transfer.gsm' (language 'en')

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

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-05-07 03:53 jamicque       New Issue                                    
2010-05-07 03:53 jamicque       Asterisk Version          => 1.6.1.19        
2010-05-07 03:53 jamicque       Regression                => No              
2010-05-07 03:53 jamicque       SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
======================================================================