[asterisk-commits] file: branch 1.2 r73466 - /branches/1.2/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Thu Jul  5 14:15:18 2007
New Revision: 73466

URL: http://svn.digium.com/view/asterisk?view=rev&rev=73466
Log:
Copy language information to the dialog structure when calling a peer for situations where a PBX may be started on the dialed channel. (issue #10121 reported by clegall_proformatique)

Modified:
    branches/1.2/channels/chan_sip.c

Modified: branches/1.2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.2/channels/chan_sip.c?view=diff&rev=73466&r1=73465&r2=73466
==============================================================================
--- branches/1.2/channels/chan_sip.c (original)
+++ branches/1.2/channels/chan_sip.c Thu Jul  5 14:15:18 2007
@@ -1915,6 +1915,8 @@
 		ast_copy_string(r->fromdomain, peer->fromdomain, sizeof(r->fromdomain));
 	if (!ast_strlen_zero(peer->fromuser))
 		ast_copy_string(r->fromuser, peer->fromuser, sizeof(r->fromuser));
+	if (!ast_strlen_zero(peer->language))
+		ast_copy_string(r->language, peer->language, sizeof(r->language));
 	r->maxtime = peer->maxms;
 	r->callgroup = peer->callgroup;
 	r->pickupgroup = peer->pickupgroup;