[asterisk-commits] mmichelson: branch group/issue8824 r172511 - /team/group/issue8824/channels/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jan 30 09:59:30 CST 2009
Author: mmichelson
Date: Fri Jan 30 09:59:30 2009
New Revision: 172511
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=172511
Log:
Fix a bad comparison and add a missing 181 case
Modified:
team/group/issue8824/channels/chan_sip.c
Modified: team/group/issue8824/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue8824/channels/chan_sip.c?view=diff&rev=172511&r1=172510&r2=172511
==============================================================================
--- team/group/issue8824/channels/chan_sip.c (original)
+++ team/group/issue8824/channels/chan_sip.c Fri Jan 30 09:59:30 2009
@@ -16842,7 +16842,7 @@
/* RFC3261 says we must treat every 1xx response (but not 100)
that we don't recognize as if it was 183.
*/
- if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && res != 181 && resp != 182 && resp != 183)
+ if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 181 && resp != 182 && resp != 183)
resp = 183;
/* Any response between 100 and 199 is PROCEEDING */
@@ -17659,6 +17659,7 @@
case 183: /* 183 Session Progress */
case 180: /* 180 Ringing */
case 182: /* 182 Queued */
+ case 181: /* 181 Call Is Being Forwarded */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
break;
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