[asterisk-commits] trunk r24565 - in /trunk: apps/app_queue.c doc/queuelog.txt

[asterisk-commits at lists.digium.com]

Author: bweschke
Date: Wed May  3 15:31:58 2006
New Revision: 24565

URL: http://svn.digium.com/view/asterisk?rev=24565&view=rev
Log:
 Log hold time and talktime in queue_log when blind transfers are made by queue members. #7038 (alphaqueue) w/documentation mods added


Modified:
    trunk/apps/app_queue.c
    trunk/doc/queuelog.txt

Modified: trunk/apps/app_queue.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?rev=24565&r1=24564&r2=24565&view=diff
==============================================================================
--- trunk/apps/app_queue.c (original)
+++ trunk/apps/app_queue.c Wed May  3 15:31:58 2006
@@ -2314,7 +2314,8 @@
 		bridge = ast_bridge_call(qe->chan,peer, &bridge_config);
 
 		if (strcasecmp(oldcontext, qe->chan->context) || strcasecmp(oldexten, qe->chan->exten)) {
-			ast_queue_log(queuename, qe->chan->uniqueid, peer->name, "TRANSFER", "%s|%s", qe->chan->exten, qe->chan->context);
+			ast_queue_log(queuename, qe->chan->uniqueid, peer->name, "TRANSFER", "%s|%s|%ld|%ld", 
+					  qe->chan->exten, qe->chan->context, (long)(callstart - qe->start), (long)(time(NULL) - callstart));
 		} else if (qe->chan->_softhangup) {
 			ast_queue_log(queuename, qe->chan->uniqueid, peer->name, "COMPLETECALLER", "%ld|%ld",
 				      (long)(callstart - qe->start), (long)(time(NULL) - callstart));

Modified: trunk/doc/queuelog.txt
URL: http://svn.digium.com/view/asterisk/trunk/doc/queuelog.txt?rev=24565&r1=24564&r2=24565&view=diff
==============================================================================
--- trunk/doc/queuelog.txt (original)
+++ trunk/doc/queuelog.txt Wed May  3 15:31:58 2006
@@ -80,7 +80,12 @@
 A call was answered by an agent, but the call was dropped because the 
 channels were not compatible.
 
-TRANSFER(extension,context)
+TRANSFER(extension|context|holdtime|calltime)
 Caller was transferred to a different extension.  Context and extension
-are recorded.
+are recorded. The caller's hold time and the length of the call are both
+recorded. PLEASE remember that transfers performed by SIP UA's by way
+of a reinvite may not always be caught by Asterisk and trigger off this
+event. The only way to be 100% sure that you will get this event when
+a transfer is performed by a queue member is to use the built-in transfer
+functionality of Asterisk.