Log hold time and talktime in queue_log when blind transfers are made by queue members. #7038 (alphaqueue) w/documentation mods added
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@24565 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -2314,7 +2314,8 @@ static int try_calling(struct queue_ent *qe, const char *options, char *announce
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bridge = ast_bridge_call(qe->chan,peer, &bridge_config);
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if (strcasecmp(oldcontext, qe->chan->context) || strcasecmp(oldexten, qe->chan->exten)) {
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ast_queue_log(queuename, qe->chan->uniqueid, peer->name, "TRANSFER", "%s|%s", qe->chan->exten, qe->chan->context);
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ast_queue_log(queuename, qe->chan->uniqueid, peer->name, "TRANSFER", "%s|%s|%ld|%ld",
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qe->chan->exten, qe->chan->context, (long)(callstart - qe->start), (long)(time(NULL) - callstart));
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} else if (qe->chan->_softhangup) {
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ast_queue_log(queuename, qe->chan->uniqueid, peer->name, "COMPLETECALLER", "%ld|%ld",
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(long)(callstart - qe->start), (long)(time(NULL) - callstart));
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@ -80,7 +80,12 @@ SYSCOMPAT
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A call was answered by an agent, but the call was dropped because the
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channels were not compatible.
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TRANSFER(extension,context)
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TRANSFER(extension|context|holdtime|calltime)
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Caller was transferred to a different extension. Context and extension
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are recorded.
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are recorded. The caller's hold time and the length of the call are both
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recorded. PLEASE remember that transfers performed by SIP UA's by way
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of a reinvite may not always be caught by Asterisk and trigger off this
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event. The only way to be 100% sure that you will get this event when
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a transfer is performed by a queue member is to use the built-in transfer
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functionality of Asterisk.
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