[asterisk-commits] mmichelson: trunk r77491 - in /trunk: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Fri Jul 27 09:31:35 2007
New Revision: 77491

URL: http://svn.digium.com/view/asterisk?view=rev&rev=77491
Log:
Merged revisions 77490 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r77490 | mmichelson | 2007-07-27 09:30:43 -0500 (Fri, 27 Jul 2007) | 3 lines

"re-invite" was misspelled


........

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?view=diff&rev=77491&r1=77490&r2=77491
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Jul 27 09:31:35 2007
@@ -12872,7 +12872,7 @@
 						if (p->vrtp)
 							ast_rtp_set_rtptimers_onhold(p->vrtp);	/* Turn off RTP timers while we send fax */
 					} else if (p->t38.state == T38_DISABLED && bridgepeer && (bridgepvt->t38.state == T38_ENABLED)) {
-						ast_log(LOG_WARNING, "RTP re-inivte after T38 session not handled yet !\n");
+						ast_log(LOG_WARNING, "RTP re-invite after T38 session not handled yet !\n");
 						/* Insted of this we should somehow re-invite the other side of the bridge to RTP */
 						/* XXXX Should we really destroy this session here, without any response at all??? */
 						sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);