[asterisk-commits] file: branch 1.4 r44450 - /branches/1.4/channels/chan_sip.c

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Wed Oct 4 19:40:41 MST 2006


Author: file
Date: Wed Oct  4 21:40:40 2006
New Revision: 44450

URL: http://svn.digium.com/view/asterisk?rev=44450&view=rev
Log:
Don't totally bail out if T.38 was negotiated

Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?rev=44450&r1=44449&r2=44450&view=diff
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Wed Oct  4 21:40:40 2006
@@ -4946,9 +4946,16 @@
 			    ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0, 0));
 	}
 	if (!newjointcapability) {
-		ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
-		/* Do NOT Change current setting */
-		return -1;
+		/* If T.38 was not negotiated either, totally bail out... */
+		if (!p->t38.jointcapability) {
+			ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
+			/* Do NOT Change current setting */
+			return -1;
+		} else {
+			if (option_debug > 2)
+				ast_log(LOG_DEBUG, "Have T.38 but no audio codecs, accepting offer anyway\n");
+			return 0;
+		}
 	}
 
 	/* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since



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