[asterisk-commits] oej: branch 1.4 r47476 - /branches/1.4/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Fri Nov 10 15:42:27 2006
New Revision: 47476

URL: http://svn.digium.com/view/asterisk?view=rev&rev=47476
Log:
...and make sure that the dialog is destroyed, even if we don't get any answer on the bye...
This is the channel that remains dead after the SIP transfer

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=47476&r1=47475&r2=47476
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Fri Nov 10 15:42:27 2006
@@ -1972,6 +1972,7 @@
 		if (option_debug > 2)
 			ast_log(LOG_DEBUG, "Finally hanging up channel after transfer: %s\n", p->callid);
 		transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
+		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 	} else
 		sip_destroy(p);
 	return 0;