Summary:	ASTERISK-09808: The language setting for Voicemail in sip.conf is not applied when the call is done with an Originate (AMI)
Reporter:	Corentin Le Gall (clegall_proformatique)	Labels:
Date Opened:	2007-07-05 12:55:47	Date Closed:	2007-07-05 14:04:07
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Applications/app_voicemail
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	This is a misfeature seen both on the last 1.2.18/19/20 versions and 1.4 branch. With the setting "language=fr" in sip.conf, the voicemail answers with the right setting when the SIP phones do call themselves "regularly". However, when the call is issued through the AMI and the Originate facility, the voicemail is left to default, ie english. ****** ADDITIONAL INFORMATION ****** When receiving a normal call, the sip_new() is called within the handle_request_invite() function, that sets te language appropriately. When issuing a call through Originate, the sip_new() is called within the sip_request_call(), which does not set this language feature at all.
Comments:	By: Digium Subversion (svnbot) 2007-07-05 13:58:43 Repository: asterisk Revision: 73466 ------------------------------------------------------------------------ r73466 | file | 2007-07-05 13:58:42 -0500 (Thu, 05 Jul 2007) | 2 lines Copy language information to the dialog structure when calling a peer for situations where a PBX may be started on the dialed channel. (issue ASTERISK-9808 reported by clegall_proformatique) ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-07-05 14:01:26 Repository: asterisk Revision: 73467 ------------------------------------------------------------------------ r73467 | file | 2007-07-05 14:01:25 -0500 (Thu, 05 Jul 2007) | 10 lines Merged revisions 73466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 lines Copy language information to the dialog structure when calling a peer for situations where a PBX may be started on the dialed channel. (issue ASTERISK-9808 reported by clegall_proformatique) ........ ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-07-05 14:03:38 Repository: asterisk Revision: 73468 ------------------------------------------------------------------------ r73468 | file | 2007-07-05 14:03:34 -0500 (Thu, 05 Jul 2007) | 18 lines Merged revisions 73467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73467 | file | 2007-07-05 16:18:02 -0300 (Thu, 05 Jul 2007) | 10 lines Merged revisions 73466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 lines Copy language information to the dialog structure when calling a peer for situations where a PBX may be started on the dialed channel. (issue ASTERISK-9808 reported by clegall_proformatique) ........ ................ ------------------------------------------------------------------------ By: Joshua C. Colp (jcolp) 2007-07-05 14:04:06 Fixed in 1.2 as of revision 73466, 1.4 as of revision 73467, and trunk as of revision 73468.