[asterisk-commits] mmichelson: branch 1.4 r77490 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Fri Jul 27 09:30:43 2007
New Revision: 77490

URL: http://svn.digium.com/view/asterisk?view=rev&rev=77490
Log:
"re-invite" was misspelled


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?view=diff&rev=77490&r1=77489&r2=77490
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Fri Jul 27 09:30:43 2007
@@ -11858,7 +11858,7 @@
 						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);