[asterisk-commits] file: branch file/issue12437 r184257 - /team/file/issue12437/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Wed Mar 25 10:36:07 2009
New Revision: 184257

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=184257
Log:
Remove some old code that can break things with T38 on the initial invite and reinvites.

Modified:
    team/file/issue12437/channels/chan_sip.c

Modified: team/file/issue12437/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/team/file/issue12437/channels/chan_sip.c?view=diff&rev=184257&r1=184256&r2=184257
==============================================================================
--- team/file/issue12437/channels/chan_sip.c (original)
+++ team/file/issue12437/channels/chan_sip.c Wed Mar 25 10:36:07 2009
@@ -12463,11 +12463,6 @@
 						ast_rtp_set_rtptimers_onhold(p->rtp);
 						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-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);
 					}
 				} else {
 					if (option_debug > 1)