Merged revisions 46937 via svnmerge from
authorKevin P. Fleming <kpfleming@digium.com>
Thu, 2 Nov 2006 16:45:50 +0000 (16:45 +0000)
committerKevin P. Fleming <kpfleming@digium.com>
Thu, 2 Nov 2006 16:45:50 +0000 (16:45 +0000)
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

channels/chan_sip.c

index eca85c4..f8b3a18 100644 (file)
@@ -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");