Re: [asterisk-users] Queue Issue : Duration between 2 agents call
Hy, I checked so many parameters that I can't see anymore a solution to this. The duration is always of 5sec. Is there a config on hardphones that could freeze the process, Does the log could help you to identify the origin of this? Regards Le 11/07/2011 09:38, Ishfaq Malik a écrit : What have you set the retry parameter for this queue? On Sun, 2011-07-10 at 13:04 +0200, Florent THOMAS wrote: Hy, I'm currently working with one queue and whatever I change in the config, it stills a gap of 6 seconds during which no agents are ringing for this queue. Is ther any parameter to configure there? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Issue : Duration between 2 agents call
Hy, I try to change the linear parameter to rrobin with memory and nothing has changed. Here is the asterisk log : /[Jul 15 14:30:05] VERBOSE[25232] app_dial.c: -- Called 5030 [Jul 15 14:30:05] VERBOSE[25230] app_queue.c: -- Local/5030@from-queue-ba73;1 is ringing [Jul 15 14:30:05] VERBOSE[25232] app_dial.c: -- SIP/5030-0065 is ringing [Jul 15 14:30:05] VERBOSE[25230] app_queue.c: -- Local/5030@from-queue-ba73;1 is ringing [Jul 15 14:30:21] VERBOSE[25230] app_queue.c: -- Nobody picked up in 15000 ms [Jul 15 14:30:21] VERBOSE[25232] app_macro.c: == Spawn extension (macro-dial-one, s, 38) exited non-zero on 'Local/5030@from-queue-ba73;2' in macro 'dial-one' [Jul 15 14:30:21] VERBOSE[25232] pbx.c: == Spawn extension (from-queue-exten-internal, 5030, 3) exited non-zero on 'Local/5030@from-queue-ba73;2' [Jul 15 14:30:21] VERBOSE[25232] pbx.c: -- Executing [h@from-queue-exten-internal:1] Macro(Local/5030@from-queue-ba73;2, hangupcall,) in new stack [Jul 15 14:30:21] VERBOSE[25232] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf(Local/5030@from-queue-ba73;2, 1?theend) in new stack [Jul 15 14:30:21] VERBOSE[25232] pbx.c: -- Goto (macro-hangupcall,s,3) [Jul 15 14:30:21] VERBOSE[25232] pbx.c: -- Executing [s@macro-hangupcall:3] Hangup(Local/5030@from-queue-ba73;2, ) in new stack [Jul 15 14:30:21] VERBOSE[25232] app_macro.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'Local/5030@from-queue-ba73;2' in macro 'hangupcall' [Jul 15 *14:30:21*] VERBOSE[25232] pbx.c: == Spawn extension (from-queue-exten-internal, h, 1) exited non-zero on 'Local/5030@from-queue-ba73;2' [Jul 15 *14:30:26*] VERBOSE[25233] pbx.c: -- Executing [5034@from-queue:1] Set(Local/5034@from-queue-0c79;2, QAGENT=5034) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [5034@from-queue:2] Goto(Local/5034@from-queue-0c79;2, 0860,1) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Goto (from-queue,0860,1) [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [0860@from-queue:1] Goto(Local/5034@from-queue-0c79;2, from-queue-exten-internal,5034,1) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Goto (from-queue-exten-internal,5034,1) [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [5034@from-queue-exten-internal:1] Set(Local/5034@from-queue-0c79;2, RingGroupMethod=none) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [5034@from-queue-exten-internal:2] Macro(Local/5034@from-queue-0c79;2, record-enable,5034,IN) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf(Local/5034@from-queue-0c79;2, 1?check) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Goto (macro-record-enable,s,4) [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf(Local/5034@from-queue-0c79;2, 0?MacroExit()) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf(Local/5034@from-queue-0c79;2, 0?Group:OUT) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Goto (macro-record-enable,s,14) [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-record-enable:14] GotoIf(Local/5034@from-queue-0c79;2, 1?IN) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Goto (macro-record-enable,s,18) [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-record-enable:18] ExecIf(Local/5034@from-queue-0c79;2, 1?MacroExit()) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [5034@from-queue-exten-internal:3] Macro(Local/5034@from-queue-0c79;2, dial-one,,trM(auto-blkvm),5034) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-dial-one:1] Set(Local/5034@from-queue-0c79;2, DEXTEN=5034) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-dial-one:2] Set(Local/5034@from-queue-0c79;2, DIALSTATUS_CW=) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-dial-one:3] GosubIf(Local/5034@from-queue-0c79;2, 0?screen,1) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-dial-one:4] GosubIf(Local/5034@from-queue-0c79;2, 0?cf,1) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-dial-one:5] GotoIf(Local/5034@from-queue-0c79;2, 1?skip1) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Goto (macro-dial-one,s,8) [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-dial-one:8] GotoIf(Local/5034@from-queue-0c79;2, 0?nodial) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-dial-one:9] GotoIf(Local/5034@from-queue-0c79;2, 0?continue) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-dial-one:10] Set(Local/5034@from-queue-0c79;2, EXTHASCW=ENABLED) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing [s@macro-dial-one:11] GotoIf(Local/5034@from-queue-0c79;2, 0?next1:cwinusebusy) in new stack [Jul 15 14:30:26] VERBOSE[25233] pbx.c: --
Re: [asterisk-users] Queue Issue : Duration between 2 agents call
Hy, I still struggle with this issue, does anybody can help me? Regards Le 10/07/2011 13:04, Florent THOMAS a écrit : Hy, I'm currently working with one queue and whatever I change in the config, it stills a gap of 6 seconds during which no agents are ringing for this queue. Is ther any parameter to configure there? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Issue : Duration between 2 agents call
What have you set the retry parameter for this queue? On Sun, 2011-07-10 at 13:04 +0200, Florent THOMAS wrote: Hy, I'm currently working with one queue and whatever I change in the config, it stills a gap of 6 seconds during which no agents are ringing for this queue. Is ther any parameter to configure there? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Issue : Duration between 2 agents call
Satish and IchFaq, thank yor for answering so fast. I checked the queue conf and the retry is put on zero : /[0860] announce-frequency=30 announce-holdtime=no announce-position=yes autofill=no eventmemberstatus=no eventwhencalled=no joinempty=yes leavewhenempty=no maxlen=0 memberdelay=0 music=default penaltymemberslimit=0 periodic-announce-frequency=0 queue-callswaiting=queue-callswaiting queue-thankyou=queue-thankyou queue-thereare=queue-thereare queue-youarenext=queue-youarenext reportholdtime=no /retry=0/ ringinuse=yes servicelevel=60 strategy=linear timeout=15 timeoutpriority=app timeoutrestart=no weight=0 wrapuptime=0 member=Local/5030@from-queue/n,0,toto member=Local/5034@from-queue/n,0,tata member=Local/5032@from-queue/n,0,titi / Le 10/07/2011 13:04, Florent THOMAS a écrit : Hy, I'm currently working with one queue and whatever I change in the config, it stills a gap of 6 seconds during which no agents are ringing for this queue. Is ther any parameter to configure there? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Issue : Duration between 2 agents call
Of course I would like to say that the problem is not solved. regards Le 11/07/2011 13:11, Florent THOMAS a écrit : Satish and IchFaq, thank yor for answering so fast. I checked the queue conf and the retry is put on zero : /[0860] announce-frequency=30 announce-holdtime=no announce-position=yes autofill=no eventmemberstatus=no eventwhencalled=no joinempty=yes leavewhenempty=no maxlen=0 memberdelay=0 music=default penaltymemberslimit=0 periodic-announce-frequency=0 queue-callswaiting=queue-callswaiting queue-thankyou=queue-thankyou queue-thereare=queue-thereare queue-youarenext=queue-youarenext reportholdtime=no /retry=0/ ringinuse=yes servicelevel=60 strategy=linear timeout=15 timeoutpriority=app timeoutrestart=no weight=0 wrapuptime=0 member=Local/5030@from-queue/n,0,toto member=Local/5034@from-queue/n,0,tata member=Local/5032@from-queue/n,0,titi / Le 10/07/2011 13:04, Florent THOMAS a écrit : Hy, I'm currently working with one queue and whatever I change in the config, it stills a gap of 6 seconds during which no agents are ringing for this queue. Is ther any parameter to configure there? regards -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Issue : Duration between 2 agents call
Hy, I'm currently working with one queue and whatever I change in the config, it stills a gap of 6 seconds during which no agents are ringing for this queue. Is ther any parameter to configure there? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Issue : Duration between 2 agents call
Check 'retry' in queues.conf [SATISH] Mumbai, India. On Sun, Jul 10, 2011 at 4:34 PM, Florent THOMAS mailingl...@tdeo.fr wrote: Hy, I'm currently working with one queue and whatever I change in the config, it stills a gap of 6 seconds during which no agents are ringing for this queue. Is ther any parameter to configure there? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue issue
It depends on what you want to do to people who are queued; if you want them to be queued, you create a queue with only one member, and have agents log on and log off as necessary; if you don't want callers to be queued, likely I would not use a queue but woul dial the agent straight. l. PS. this is quite an unusual requirement, what is it for? 2009/9/1 Paul Hales pdha...@optusnet.com.au I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue issue
A situation where staff want a mobile and their SIP handset to share an extension - but to make sure the mobile or SIP handset do not ring if they are speaking on the other one... PaulH Lenz Emilitri wrote: It depends on what you want to do to people who are queued; if you want them to be queued, you create a queue with only one member, and have agents log on and log off as necessary; if you don't want callers to be queued, likely I would not use a queue but woul dial the agent straight. l. PS. this is quite an unusual requirement, what is it for? 2009/9/1 Paul Hales pdha...@optusnet.com.au mailto:pdha...@optusnet.com.au I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue issue
One way to do this would be to use hints and an AGI to control dialing.
Let's say you have extensions 100 and 101 and each staffer also has a cell
(555-1212 and 555-1213). When you dial 100, you want to ring 100 and
555-1212 if both are available and the same with 101 and 555-1213. This
snippet would do it:
- exten = s,1XX,Macro(ring-group,${EXTEN})
- exten = s,1XX,playback(vm-goodbye)
- exten = s,1XX,hangup
- [macro-ring-group]
- exten = s,1,AGI(checkhints.agi,${ARG1})
- exten = s,n,gotoif($[${LINESTAT} = BUSY]?inuse)
- exten = s,n,Dial(SIP/${ARG1}DAHDI/g1/${CELLLINE},60)
- exten = s,n,hangup
- exten = s,n(inuse),playback(line-in-use)
- exten = s,n,hangup
The AGI checks the hint for 100 or 101 and assigns CELLLINE to call the
cell. If either is in use, LINESTAT is set to BUSY, otherwise set to AVAIL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hales
Sent: Wednesday, September 02, 2009 2:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue issue
A situation where staff want a mobile and their SIP handset to share an
extension - but to make sure the mobile or SIP handset do not ring if
they are speaking on the other one...
PaulH
Lenz Emilitri wrote:
It depends on what you want to do to people who are queued; if you
want them to be queued, you create a queue with only one member, and
have agents log on and log off as necessary; if you don't want callers
to be queued, likely I would not use a queue but woul dial the agent
straight.
l.
PS. this is quite an unusual requirement, what is it for?
2009/9/1 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
I have a _very_ specific situation where I need queues to work in
a very
specific manner - I need the queue to only accept one call at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this way in the
distant past (pre 1.0)...but I am willing to be mistaken.
Is this even remotely possible?
PaulH
--
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] queue issue
On 3/09/09 11:34 AM, Paul Hales wrote: Hmmm.any idea how I can use hints to monitor their mobile phones? Unless the call came in via Asterisk, you can't. Why not just have the desk phone accept one call (i.e. call/group/whatever limit) and then use app_followme? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue issue
Matt Riddell wrote: On 3/09/09 11:34 AM, Paul Hales wrote: Hmmm.any idea how I can use hints to monitor their mobile phones? Unless the call came in via Asterisk, you can't. The calls will - so it should be able (at the very least with the asterisk internal DB - which I don't fully trust due to reboots and the odd weird behaviour) Why not just have the desk phone accept one call (i.e. call/group/whatever limit) and then use app_followme? The issue is that both phones have to ring at the same time.And it's easy enough to stop the mobile from ringing if the SIP phone is in use, but the other way around is the challengeIt's doable, but I want to find the right solution. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue issue
They don't want to log in, and they want both to ring if they are free -
this is a very large site, so they need to be contactable at all times.
PaulH
Lenz Emilitri wrote:
I would have them log on with the mobile when they need it, and log
off when they don't. When the mobile is not present you would simply
dial the local extension.
You could have something like:
local/1...@agents
that does something like:
if ( DBSET(has_mobile) ) {
dial( Zap/g0/MYMOBILENUM )
} else {
dial( SIP/123 )
}
and have anothe extension set/reset the has_mobile property in the AstDB.
You could then call Local/1...@gaents directkly or make it a member of
the queue (with known issues on some version of *) :-)
l.
2009/9/2 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
A situation where staff want a mobile and their SIP handset to
share an
extension - but to make sure the mobile or SIP handset do not ring if
they are speaking on the other one...
PaulH
Lenz Emilitri wrote:
It depends on what you want to do to people who are queued; if you
want them to be queued, you create a queue with only one member, and
have agents log on and log off as necessary; if you don't want
callers
to be queued, likely I would not use a queue but woul dial the agent
straight.
l.
PS. this is quite an unusual requirement, what is it for?
2009/9/1 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au mailto:pdha...@optusnet.com.au
I have a _very_ specific situation where I need queues to
work in
a very
specific manner - I need the queue to only accept one call
at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this
way in the
distant past (pre 1.0)...but I am willing to be mistaken.
Is this even remotely possible?
PaulH
--
Loway - home of QueueMetrics - http://queuemetrics.com
___
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http://www.api-digital.com --
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Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Loway - home of QueueMetrics - http://queuemetrics.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
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Re: [asterisk-users] queue issue
Hmmm.any idea how I can use hints to monitor their mobile phones?
PaulH
Danny Nicholas wrote:
One way to do this would be to use hints and an AGI to control dialing.
Let's say you have extensions 100 and 101 and each staffer also has a cell
(555-1212 and 555-1213). When you dial 100, you want to ring 100 and
555-1212 if both are available and the same with 101 and 555-1213. This
snippet would do it:
- exten = s,1XX,Macro(ring-group,${EXTEN})
- exten = s,1XX,playback(vm-goodbye)
- exten = s,1XX,hangup
- [macro-ring-group]
- exten = s,1,AGI(checkhints.agi,${ARG1})
- exten = s,n,gotoif($[${LINESTAT} = BUSY]?inuse)
- exten = s,n,Dial(SIP/${ARG1}DAHDI/g1/${CELLLINE},60)
- exten = s,n,hangup
- exten = s,n(inuse),playback(line-in-use)
- exten = s,n,hangup
The AGI checks the hint for 100 or 101 and assigns CELLLINE to call the
cell. If either is in use, LINESTAT is set to BUSY, otherwise set to AVAIL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hales
Sent: Wednesday, September 02, 2009 2:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue issue
A situation where staff want a mobile and their SIP handset to share an
extension - but to make sure the mobile or SIP handset do not ring if
they are speaking on the other one...
PaulH
Lenz Emilitri wrote:
It depends on what you want to do to people who are queued; if you
want them to be queued, you create a queue with only one member, and
have agents log on and log off as necessary; if you don't want callers
to be queued, likely I would not use a queue but woul dial the agent
straight.
l.
PS. this is quite an unusual requirement, what is it for?
2009/9/1 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
I have a _very_ specific situation where I need queues to work in
a very
specific manner - I need the queue to only accept one call at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this way in the
distant past (pre 1.0)...but I am willing to be mistaken.
Is this even remotely possible?
PaulH
--
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] queue issue
On 3/09/09 12:21 PM, Paul Hales wrote: Matt Riddell wrote: On 3/09/09 11:34 AM, Paul Hales wrote: Hmmm.any idea how I can use hints to monitor their mobile phones? Unless the call came in via Asterisk, you can't. The calls will - so it should be able (at the very least with the asterisk internal DB - which I don't fully trust due to reboots and the odd weird behaviour) Then it's easy :) Use func_devstate - you can set custom device states for things - and btw the Asterisk DB is pretty stable - we're using it in pretty large call centres without (touch wood) ever having any problems. A lot more than I can say for MySQL :) Oh, by the way, func_devstate was only added to 1.4 a few months back - although if you're stuck with a particular version, the backport always applied cleanly for me. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue issue
On Tue, Sep 1, 2009 at 4:35 AM, Paul Halespdha...@optusnet.com.au wrote: Miguel Molina wrote: Paul Hales escribió: I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH Hi, Maybe maxlen = 1? Cheers, Hmmm - almost. Maxlen limits the amounts of calls waiting for the queue, not the amount of callers talking to queue members. You can do any limitations i can imagine with Set(GROUP()=...) and GROUP_COUNT. Do You actually need rest of callers to wait in queue while one is speaking, or disconnect them before they enter queue? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue issue
I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue issue
Paul Hales escribió: I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, Maybe maxlen = 1? Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue issue
Miguel Molina wrote: Paul Hales escribió: I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH Hi, Maybe maxlen = 1? Cheers, Hmmm - almost. Maxlen limits the amounts of calls waiting for the queue, not the amount of callers talking to queue members. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Issue (1.4.21.1)
What version do you mean.. 1.6? Upgrading might be a option, but we cant loose any functionality/stability - Original Message - From: Paul Hales [mailto:pdha...@optusnet.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 14:57:26 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) I think the handling of this may have improved in later versions of Asterisk - is an upgrade an option? (I tested this with a newer version of Asterisk recently, and it behaved how you were hoping it would behave) PaulH Kev Szaszvari wrote: The strange thing is, Queue calls are working as per expected. If they get a call from the queue they wont get another until the 1st call is done. Its only when the agent received a direct call or a internal call from another staff member, the queue continues to ring their phone. - Original Message - From: Kev Szaszvari [mailto:k...@mailcall.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:36:32 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) It appears that that option is set from queues.conf [ops] musicclass = default strategy = leastrecent timeout = 5 retry = 1 wrapuptime= 3 autofill = yes autopause = no maxlen = 0 joinempty = yes leavewhenempty = no ringinuse = no - Original Message - From: Paul Hales [mailto:pdha...@optusnet.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:01:40 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information
[asterisk-users] Queue Issue (1.4.21.1)
Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Issue (1.4.21.1)
The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Issue (1.4.21.1)
It appears that that option is set from queues.conf [ops] musicclass = default strategy = leastrecent timeout = 5 retry = 1 wrapuptime= 3 autofill = yes autopause = no maxlen = 0 joinempty = yes leavewhenempty = no ringinuse = no - Original Message - From: Paul Hales [mailto:pdha...@optusnet.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:01:40 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Issue (1.4.21.1)
The strange thing is, Queue calls are working as per expected. If they get a call from the queue they wont get another until the 1st call is done. Its only when the agent received a direct call or a internal call from another staff member, the queue continues to ring their phone. - Original Message - From: Kev Szaszvari [mailto:k...@mailcall.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:36:32 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) It appears that that option is set from queues.conf [ops] musicclass = default strategy = leastrecent timeout = 5 retry = 1 wrapuptime= 3 autofill = yes autopause = no maxlen = 0 joinempty = yes leavewhenempty = no ringinuse = no - Original Message - From: Paul Hales [mailto:pdha...@optusnet.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:01:40 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Issue (1.4.21.1)
I think the handling of this may have improved in later versions of Asterisk - is an upgrade an option? (I tested this with a newer version of Asterisk recently, and it behaved how you were hoping it would behave) PaulH Kev Szaszvari wrote: The strange thing is, Queue calls are working as per expected. If they get a call from the queue they wont get another until the 1st call is done. Its only when the agent received a direct call or a internal call from another staff member, the queue continues to ring their phone. - Original Message - From: Kev Szaszvari [mailto:k...@mailcall.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:36:32 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) It appears that that option is set from queues.conf [ops] musicclass = default strategy = leastrecent timeout = 5 retry = 1 wrapuptime= 3 autofill = yes autopause = no maxlen = 0 joinempty = yes leavewhenempty = no ringinuse = no - Original Message - From: Paul Hales [mailto:pdha...@optusnet.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:01:40 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want
[Asterisk-Users] queue issue
Hi, I have several queues configured at my call center for different support levels. Today, something weird happened: - A client called queue 1 and was answered by an agent - The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf - The user transferred the client to another Queue, by using the second channel and the XFer key of her EyeBeam softphone) - The client entered this second queue and was answered correctly by an analyst from this second queue. But, when I ran "show queue secondqueue" or "show agents", even though the analyst is busy, she appear as available and the call is not registered in queue_log or anywhere else. She also can receive other calls from this queue, since she is not considered busy by the Queue application. Has anybody already realized this issue? Is this a bug or a misuse? Thank you!!!Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue issue
On 04/05/06 21:37 Dov Bigio said the following: - The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf on a related note, we notice that if we've set atxfer = *1 in features.conf and blindxfer=#1, then attended transfers dont work. somehow, the Queue app captures the '*' and hangs up the call. is this the behaviour others have observed ? obviously, since we've used *2 for auto monitor, that doesnt work as well. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
