[asterisk-commits] mmichelson: branch 1.6.2 r249104 - in /branches/1.6.2: ./ channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Feb 26 11:06:07 CST 2010


Author: mmichelson
Date: Fri Feb 26 11:06:02 2010
New Revision: 249104

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=249104
Log:
Merged revisions 249101 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb 2010) | 14 lines
  
  Merged revisions 249100 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb 2010) | 8 lines
    
    For T.38 reINVITEs treat a 606 the same as a 488.
    
    (closes issue #16792)
    Reported by: vrban
    Patches:
          t38_606.patch uploaded by vrban (license 756)
  ........
................

Modified:
    branches/1.6.2/   (props changed)
    branches/1.6.2/channels/chan_sip.c

Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=249104&r1=249103&r2=249104
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Fri Feb 26 11:06:02 2010
@@ -17832,6 +17832,7 @@
 		break;
 	case 415: /* Unsupported media type */
 	case 488: /* Not acceptable here */
+	case 606: /* Not Acceptable */
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
 		if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
 			change_t38_state(p, T38_DISABLED);
@@ -18471,6 +18472,7 @@
 			break;
 		case 415: /* Unsupported media type */
 		case 488: /* Not acceptable here - codec error */
+		case 606: /* Not Acceptable */
 			if (sipmethod == SIP_INVITE)
 				handle_response_invite(p, resp, rest, req, seqno);
 			break;




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