Merged revisions 73467 via svnmerge from
authorJoshua Colp <jcolp@digium.com>
Thu, 5 Jul 2007 19:20:12 +0000 (19:20 +0000)
committerJoshua Colp <jcolp@digium.com>
Thu, 5 Jul 2007 19:20:12 +0000 (19:20 +0000)
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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 #10121 reported by clegall_proformatique)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73468 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index cfc79c9..4ade39e 100644 (file)
@@ -3056,6 +3056,8 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
                ast_string_field_set(dialog, fromdomain, peer->fromdomain);
        if (!ast_strlen_zero(peer->fromuser))
                ast_string_field_set(dialog, fromuser, peer->fromuser);
+       if (!ast_strlen_zero(peer->language))
+               ast_string_field_set(dialog, language, peer->language);
        dialog->callgroup = peer->callgroup;
        dialog->pickupgroup = peer->pickupgroup;
        dialog->allowtransfer = peer->allowtransfer;