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Merged revisions 46937 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r46937 | kpfleming | 2006-11-02 10:45:32 -0600 (Thu, 02 Nov 2006) | 2 lines

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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
2006-11-02 16:45:50 +00:00
parent aebb4aaa0d
commit 30b47fb8a0
1 changed files with 6 additions and 0 deletions
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6
channels/chan_sip.c
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@@ -6096,6 +6096,12 @@ 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");