[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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