[svn-commits] trunk r24565 - in /trunk: apps/app_queue.c
doc/queuelog.txt
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Wed May 3 13:31:58 MST 2006
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.
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