[asterisk-commits] file: trunk r44451 - in /trunk: ./ channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: file
Date: Wed Oct  4 21:43:07 2006
New Revision: 44451

URL: http://svn.digium.com/view/asterisk?rev=44451&view=rev
Log:
Merged revisions 44450 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r44450 | file | 2006-10-04 22:40:40 -0400 (Wed, 04 Oct 2006) | 2 lines

Don't totally bail out if T.38 was negotiated

........

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=44451&r1=44450&r2=44451&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Oct  4 21:43:07 2006
@@ -4942,9 +4942,16 @@
 			    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