From efdffc0cf8168a2c3d479a7efcb465fecba17487 Mon Sep 17 00:00:00 2001 From: "Kevin P. Fleming" Date: Thu, 2 Nov 2006 16:45:32 +0000 Subject: [PATCH] don't send INVITE when we have determined that we can't offer any audio formats due to lack of trancoding support (or incorrect configuration) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46937 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index aad6b0c80e..3a59655619 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -6035,6 +6035,12 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p) /* Ok, let's start working with codec selection here */ capability = ast_translate_available_formats(p->jointcapability, p->prefcodec); + /* If there are no audio formats left to offer, punt */ + if (!(capability & AST_FORMAT_AUDIO_MASK)) { + ast_log(LOG_WARNING, "No audio format found to offer.\n"); + return -1; + } + if (option_debug > 1) { char codecbuf[BUFSIZ]; ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False");