From 44a9af35761c45010dc320c4786d683bdadaefb3 Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Thu, 1 Feb 2007 22:26:11 +0000 Subject: [PATCH] Merged revisions 53104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53104 | file | 2007-02-01 16:24:32 -0600 (Thu, 01 Feb 2007) | 10 lines Merged revisions 53103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index da60920561..4767ac6a16 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -2961,7 +2961,8 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) p->callingpres = ast->cid.cid_pres; p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec); - + p->jointnoncodeccapability = p->noncodeccapability; + /* If there are no audio formats left to offer, punt */ if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) { ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);