search for: setqueueentryvar

...eues, it would be easy because I would run the netcat command and then dial the user's extension. My Question is: Is there a way I can know when the caller is just about to be connected to an agent (when the agent's SIP extension starts ringing)? There are these settings setinterfacevar, setqueueentryvar, setqueuevar in queues.conf but when can I use them? Have you guys been in this situation before? Any alternative solutions (sending caller info to an agent)? I am using Asterisk 11 and Windows 7 PCs for agents. Thank you! Kind Regards, Wilson

Hi, There are 3 new settings (setinterfacevar, setqueueentryvar, setqueuevar) and membermacro settings in 1.6 queues.conf. What is the potential use of these settings? The variables set are useful, but there is no indication of the purpose they could be used? Any one with some light on potential use case of these new features? raj

...agent answers the phone. When the local channel is used, the queue related variables, specifically MEMBERINTERFACE, are missing. When a normal call (typically SIP or DAHDI channel) enters the queue, the MEMBERINTERFACE and other variables are present. my queues.conf has setinterfacevar = yes setqueueentryvar = yes setqueuevar = yes ; I didn't see anything in the V12 doc that related to this. Is this a bug or a feature? -- Mitch

...nswer exten => _X.,2,Dial(SIP/{EXTEN},20,tT) queues.conf [general] persistentmembers = yes autofill = yes Canreinvite=yes ; (tried with NO also) monitor-type = MixMonitor [55365] fullname = Frontdesk strategy = roundrobin context=from-internal ringinuse=no setinterfacevar=yes setqueueentryvar=yes timeout = 10 wrapuptime = autofill = yes autopause = no maxlen = joinempty = no leavewhenempty = no reportholdtime = no musicclass = call-limit = 20 member = SIP/100 member = SIP/101 member = SIP/102 Please help , I m in a total mess .Thanks Sriram -------------- next part -------...

...nuse=no ; don't ring members when already InUse (prevents multiple calls to an agent) announce-frequency = 30 min-announce-frequency = 15 announce-holdtime = yes|no|once announce-position = limit announce-position-limit = 5 announce-round-seconds = 10 setinterfacevar = yes ; set some variables setqueueentryvar = yes ; some more variables setqueuevar = yes ; some more variables wrapuptime=2 ; wait 2 seconds before next call timeout = 16 ; try each operator for 12 seconds before moving to the next one autopause=no ; don't pause a member when they fail to answer a call [sales](StandardQueue) ; create...

Is there any option in Asterisk distribution of this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120531/8fac6a22/attachment.htm>

...risk Open Source PBX # grep -vE '^\s*;.*|^\s*$' queues.conf [general] persistentmembers = yes monitor-type = MixMonitor [defaultq] musicclass = default_noannounce announce = tt-busy strategy = ringall context = qoutcon timeout = 10 retry = 99 wrapuptime=15 maxlen = 0 setinterfacevar=yes setqueueentryvar=yes setqueuevar=yes announce-frequency = 30 min-announce-frequency = 30 periodic-announce-frequency = 30 announce-holdtime = yes announce-position = yes announce-to-first-user = yes periodic-announce = tt-allbusy,hold-or-dial-0 monitor-format = wav49 monitor-type = MixMonitor reportholdtime = yes...

I enabled the logs on the mysql database and ran : realtime load queues name cou0002-test in the mysql log I can see that the proper select statement is being executed: 2018-12-04T16:29:27.253094Z 229 Query SET SESSION TRANSACTION ISOLATION LEVEL READ COMMITTED 2018-12-04T16:29:27.254384Z 229 Prepare SELECT * FROM queues WHERE name = ? 2018-12-04T16:29:27.254902Z 229

...n instead of context, without providing for the Referred-To SIP URI (Reported by Matt Jordan) * ASTERISK-24513 - Local channel apparently leaked in off-nominal DTMF attended transfer (Reported by Mark Michelson) * ASTERISK-24267 - Queue variables associated with setinterfacevar, setqueueentryvar, setqueuevar are not passed to local channel (Reported by Mitch Claborn) * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall calls to the transferrer. (Reported by Richard Mudgett) * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong destination when...

...SK-24513 - Local channel apparently leaked in off-nominal DTMF attended transfer (Reported by Mark Michelson) * ASTERISK-24367 - PJSIP: allow all results in failure to send INVITE (Reported by Scott Griepentrog) * ASTERISK-24267 - Queue variables associated with setinterfacevar, setqueueentryvar, setqueuevar are not passed to local channel (Reported by Mitch Claborn) * ASTERISK-24641 - Deadlock in Trunk (Reported by Malcolm Davenport) * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall calls to the transferrer. (Reported by Richard Mudgett) * ASTERIS...

...24367 <https://issues.asterisk.org/jira/browse/ASTERISK-24367>] - PJSIP: allow all results in failure to send INVITE (Reported by Scott Griepentrog) - [ASTERISK-24267 <https://issues.asterisk.org/jira/browse/ASTERISK-24267>] - Queue variables associated with setinterfacevar, setqueueentryvar, setqueuevar are not passed to local channel (Reported by Mitch Claborn) - [ASTERISK-24641 <https://issues.asterisk.org/jira/browse/ASTERISK-24641>] - Deadlock in Trunk (Reported by Malcolm Davenport) - [ASTERISK-23841 <https://issues.asterisk.org/jira/browse/ASTERISK-23841&...