[asterisk-commits] mmichelson: branch group/v14_colp r158379 - /team/group/v14_colp/channels/

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Fri Nov 21 11:25:54 2008
New Revision: 158379

URL: http://svn.digium.com/view/asterisk?view=rev&rev=158379
Log:
Add parsing of connected line information on incoming
200 OK responses to an INVITE. While it's not likely that
this will be needed, I could see some inventive setups where
some programmable proxy never sends a provisional response to
an INVITE but does send a 200 OK with identifying information.


Modified:
    team/group/v14_colp/channels/chan_sip.c

Modified: team/group/v14_colp/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/group/v14_colp/channels/chan_sip.c?view=diff&rev=158379&r1=158378&r2=158379
==============================================================================
--- team/group/v14_colp/channels/chan_sip.c (original)
+++ team/group/v14_colp/channels/chan_sip.c Fri Nov 21 11:25:54 2008
@@ -12901,6 +12901,16 @@
 					/* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
 					/* For re-invites, we try to recover */
 					ast_set_flag(&p->flags[0], SIP_PENDINGBYE);	
+		}
+
+		if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner && get_rpid(p, req)) {
+			/* Queue a connected line update */
+			ast_party_connected_line_init(&connected);
+			connected.id.number = (char *) p->cid_num;
+			connected.id.name = (char *) p->cid_name;
+			connected.id.number_presentation = p->callingpres;
+			connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
+			ast_queue_connected_line_update(p->owner, &connected);
 		}
 
 		/* Parse contact header for continued conversation */