Re: [asterisk-users] Queue Timeout

Hi, To answer your questions. 1. Not sure on statically configured queues. This is a production system that I can't break down currently to test. 2. Our member table has the following fields, uniqueid, member name, queue name, interface, penalty, paused. Which are 425, Nick Test Phone,

[asterisk-users] Queue Timeout

Hello, We're running asterisk 16 with Realtime. We have queues configured in realtime. The "Timeout" setting appears to have an upper 2 minute limit. Even when setting the timeout in the queue to 600 seconds, the agent is no longer rung after exactly 120 seconds. The asterisk CLI claims "Exiting

Re: [asterisk-users] Queue don't call Interface PJSIP

[SOLVED]!!! My function that changed the callerid was returning an invalid number. Although the asterisk sends the call, the SIP header was wrong and the extension did not ring Thanks. Em 18/08/2020 09:07, Joshua C. Colp escreveu: On Tue, Aug 18, 2020 at 9:00 AM Roberto

Re: [asterisk-users] Queue don't call Interface PJSIP

On Tue, Aug 18, 2020 at 9:00 AM Roberto < roberto.med...@gasparimsantos.com.br> wrote: > Hi Joshua, thanks for answer. > In this particular test my extension is on a simple network. There is no > NAT, just an asterisk running on a virtual machine on a 7- 64bit CentOs. I > am simulating an

Re: [asterisk-users] Queue don't call Interface PJSIP

Hi Joshua, thanks for answer. In this particular test my extension is on a simple network. There is no NAT, just an asterisk running on a virtual machine on a 7- 64bit CentOs. I am simulating an environment to be able to use PJSIP on my client. And even in this small environment, my extension

Re: [asterisk-users] Queue don't call Interface PJSIP

On Mon, Aug 17, 2020 at 6:16 PM Roberto < roberto.med...@gasparimsantos.com.br> wrote: > Hello. > > > I am having a lot of problems with SIP through NAT. So, I decided to adopt > PJSIP. However, I am not able to make the extensions ring when receiving a > call from the queue. I'm using telnet to

[asterisk-users] Queue don't call Interface PJSIP

Hello. I am having a lot of problems with SIP through NAT. So, I decided to adopt PJSIP. However, I am not able to make the extensions ring when receiving a call from the queue. I'm using telnet to include the extension and on the asterisk console, it even shows Called PJSIP/6001, but the

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

0 > retry=1 > > Which means if I send it to "Eric" - it will go to his voicemail after 30 > seconds. Should I change timings? > > Thank you! > > -- > *From:* John Kiniston > *To:* Ivan Demkovitch ; Asterisk Users Mailing > List - Non-Commercial Discussion >

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

: [asterisk-users] Queue not dialing out to cell phone for some reason So, LOCAL in this context is a 'Technology' or 'Channel Driver' , Instead of PJSIP, SIP, IAX, it's sending a call to a dialplan target. Your entry in queues.conf with LOCAL/105@internal would send the call to the context

Re: [asterisk-users] Queue member not local - PJSIP - Asterisk 16

No one on this ? Le 22/11/2018 à 17:59, Administrator TOOTAI a écrit : Hi all, I want to set dynamic queue with non local members. I create an extension 115 in [localEP] context which is doing the job, eg calls to this extension are forwarded to the non local endpoint (which is an IP phone

[asterisk-users] Queue member not local - PJSIP - Asterisk 16

Hi all, I want to set dynamic queue with non local members. I create an extension 115 in [localEP] context which is doing the job, eg calls to this extension are forwarded to the non local endpoint (which is an IP phone connected to an external Asterisk 13 version). Phones are SNOM. Queue

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

e timings? Thank you! From: John Kiniston To: Ivan Demkovitch ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, November 16, 2018 2:43 PM Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some reason My settings for the queue

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

> security,notice,warning,error,fax > verbose => verbose > > > > Thank you! > > ------ > *From:* John Kiniston > *To:* idemkovi...@yahoo.com > *Sent:* Thursday, November 15, 2018 3:17 PM > *Subject:* Re: [asterisk-users] Queue not dialin

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

al] dateformat=%F %T [logfiles] console => notice,warning,error,dtmf messages => security,notice,warning,error,fax verbose => verbose Thank you! From: John Kiniston To: idemkovi...@yahoo.com Sent: Thursday, November 15, 2018 3:17 PM Subject: Re: [asterisk-users] Queue

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

From: John Kiniston To: idemkovi...@yahoo.com Sent: Thursday, November 15, 2018 3:17 PM Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some reason OK. So it looks like asterisk can't ring FF1565AABB2D-SLS because it's invalid. is the user at  '155

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

ber 15, 2018 2:21 PM Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some reason what does the output of 'queue show sales' show? Do you have queue logging enabled? Have you looked in the queue log to see what events are firing? On Thu, Nov 15, 2018 at 9:55 AM Ivan Demkovit

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

what does the output of 'queue show sales' show? Do you have queue logging enabled? Have you looked in the queue log to see what events are firing? On Thu, Nov 15, 2018 at 9:55 AM Ivan Demkovitch wrote: > Hello, > > I have queues.conf setup with a group like so: > > [Sales](StandardQueue) >

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

of ”Registred” to your trunk operator. Från: Ivan Demkovitch Skickat: den 15 november 2018 18:01 Till: Sebastian Nielsen ; 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: Re: SV: [asterisk-users] Queue not dialing out to cell phone for some reason Sebastian, I don't

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

Users Mailing List - Non-Commercial Discussion' Sent: Thursday, November 15, 2018 10:58 AM Subject: SV: [asterisk-users] Queue not dialing out to cell phone for some reason #yiv7898733751 #yiv7898733751 -- _filtered #yiv7898733751 {font-family:Helvetica;panose-1:2 11 6 4 2 2 2 2 2 4

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

into android settings and make sure the SIP client is whitelisted in battery management. Från: asterisk-users För Ivan Demkovitch Skickat: den 15 november 2018 17:55 Till: asterisk-users@lists.digium.com Ämne: [asterisk-users] Queue not dialing out to cell phone for some reason Hello, I

[asterisk-users] Queue not dialing out to cell phone for some reason

Hello, I have queues.conf setup with a group like so: [Sales](StandardQueue) announce = first member => SIP/FF4C119EEBF8-SLS member => SIP/FF9EF375CCFC-SLS member => SIP/1314555@callcentric ;Eric's cell member => SIP/FF1565AABB2D-SLS ;Eric's Yealink So, my idea here that it should ring all 4

Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

> It does seem like a bug. However, you have a complicated dialplan with a lot > of pieces happening at > once so it may not actually be an Asterisk bug but a problem with your > dialplan. To unravel this is > going to take some bookkeeping on your part. Hi Richard, Thanks for the detailed

Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

On Wed, Aug 8, 2018 at 7:43 PM, Daniel Journo wrote: > > Doing some more tests, this reads like a bug to me. > > Using a hanguphandler with DumpChan in the dialplan context that executes > > the Queue, I can see that DYNAMIC_FEATURES is set. > > After the attended transfer when the call is

Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

> Doing some more tests, this reads like a bug to me. > Using a hanguphandler with DumpChan in the dialplan context that executes > the Queue, I can see that DYNAMIC_FEATURES is set. > After the attended transfer when the call is ended, the hanguphandler still > shows that DYNAMIC_FEATURES is set.

Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

On Wed, Aug 8, 2018 at 1:53 PM, Daniel Journo wrote: > > Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. > > AgentA answers and is able to use that feature code. > > If AgentA performs an attended transfer of a call from a queue to > AgentB, the > > feature code no

Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

> Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. > AgentA answers and is able to use that feature code. > If AgentA performs an attended transfer of a call from a queue to AgentB, the > feature code no longer works. > > It only doesn't work when using Queue() and an

[asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

Hi, I think I've identified an issue and just want to check before completing a bug report. Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. AgentA answers and is able to use that feature code. If AgentA performs an attended transfer of a call from a queue to AgentB,

[asterisk-users] Queue of automated members

We're using Asterisk 14.7.6, and we're able to route particular extensions to a corresponding automated member via SIP. Each extension has its own specific content, and all the automated members are configured to handle any of these extensions. We would like to achieve some scaling by putting all

[asterisk-users] Queue playing periodic_announce to agent when they answer

    I have a very strange problem with my queues today.  When the agent answers a call they get the periodic_announce sound played to them.  I have a periodic_announce set to 60 seconds and the caller does hear it if their call is not answered.  Why would it play it to the agent?  At this

Re: [asterisk-users] queue peridiodic-announce-frequency

On Wed, 17 Jan 2018 12:08:40 +0100 Antony Stone wrote: > On Wednesday 17 January 2018 at 11:59:21, Paul Neuwirth wrote: > > > Hello group, > > > > I tried a lot to enlarge the frequency (i.e. more announces, low > > wait between). according to config,

Re: [asterisk-users] queue peridiodic-announce-frequency

On Wednesday 17 January 2018 at 11:59:21, Paul Neuwirth wrote: > Hello group, > > I tried a lot to enlarge the frequency (i.e. more announces, low wait > between). according to config, every 30 seconds the announcement should > take place. In fact, the first periodic announce is done after 2 >

[asterisk-users] queue peridiodic-announce-frequency

Hello group, I tried a lot to enlarge the frequency (i.e. more announces, low wait between). according to config, every 30 seconds the announcement should take place. In fact, the first periodic announce is done after 2 minutes? What is my fault? Thank you Regards Paul # zypper if asterisk

[asterisk-users] Queue, no announcement being played at all

Hi List I have a very strange problem. I was using queues a while ago with an asterisk 1.2 or so and announcements were working fine more or less out of the box. Now I am once again trying to set up a queue with Version 13.14.1 an not matter what I do, I don't get the announcements to be played.

[asterisk-users] queue show - extensions in call going from (in use) to (not in use)

Running Asterisk 11.23.0 realtime currently, I've noticed some odd queue behavior only starting today. We are using queues in conjunction with FOP2 to show agent status, etc. As of today, we noticed that an agent will be on a call but show as available. On an inbound queue call, when an agent

Re: [asterisk-users] Queue show : failed to extend from 240 to 327

On Sat, Sep 10, 2016 at 5:18 AM, Jonas Kellens wrote: > On 10-09-16 09:42, Jonas Kellens wrote: > > > On 10-09-16 00:50, Richard Mudgett wrote: > > > > On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens > wrote: > >> Hello >> >> when I type on the

Re: [asterisk-users] Queue show : failed to extend from 240 to 327

On 10-09-16 09:42, Jonas Kellens wrote: On 10-09-16 00:50, Richard Mudgett wrote: On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens > wrote: Hello when I type on the Asterisk CLi 'queue show', I first get a list of my

Re: [asterisk-users] Queue show : failed to extend from 240 to 327

On 10-09-16 00:50, Richard Mudgett wrote: On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens > wrote: Hello when I type on the Asterisk CLi 'queue show', I first get a list of my queues and then the following : failed

Re: [asterisk-users] Queue show : failed to extend from 240 to 327

On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens wrote: > Hello > > when I type on the Asterisk CLi 'queue show', I first get a list of my > queues and then the following : > > > failed to extend from 240 to 327 > failed to extend from 240 to 334 > > > I could not find

[asterisk-users] Queue show : failed to extend from 240 to 327

Hello when I type on the Asterisk CLi 'queue show', I first get a list of my queues and then the following : failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 323 failed to extend from 240 to 327 failed to extend

Re: [asterisk-users] Queue grouping - how can it be implemented?

Use Local Channels and hints to combine SIP/MOM and SIP/MOMMobile into a single extension you add to the queue. extensions.conf: [queue-phones] exten => MOM,1,Dial(SIP/MOM/MOMMOBILE,60,tkw) exten => MOM,hint,SIP/MOM/MOMMOBILE exten => DAD,1,Dial(SIP/DAD/DADMOBILE,60,tkw) exten =>

[asterisk-users] Queue grouping - how can it be implemented?

I have a Asterisk set up. In this, I want to use queues. Now I want to group "agents" into groups, such as so if one phone in a group is busy, the whole group is considered busy. Eg: Group1: SIP/Dad SIP/DadsMobile Group2: SIP/Mom SIP/MomsMobile If there is three persons in

[asterisk-users] Queue logfile txt format in mySQL needed

Hello, Iam using queues and agents, thats OK. I have interesting information form Asterisk in txt file format var/log/asterisk/queue_log Today Iam reading these txt files and wrote them in an mySQL databases. I would need this information more realtime. Some information I do writing in the

Re: [asterisk-users] Queue logfile txt format in mySQL needed

> From: Thomas <thomasit...@gmail.com> > To: asterisk-users@lists.digium.com, > Date: 01/21/2016 04:17 AM > Subject: [asterisk-users] Queue logfile txt format in mySQL needed > Sent by: asterisk-users-boun...@lists.digium.com > > Hello, > > Iam using queues

Re: [asterisk-users] Queue logfile txt format in mySQL needed

Am Donnerstag, 21. Januar 2016, 09:52:53 schrieben Sie: > > From: Thomas <thomasit...@gmail.com> > > To: asterisk-users@lists.digium.com, > > Date: 01/21/2016 04:17 AM > > Subject: [asterisk-users] Queue logfile txt format in mySQL needed > > Sent by: ast

[asterisk-users] Queue priority not respected

Hello I notice that priority of queue members is not being respected. Using mysql realtime. These are the queue members (in table queue_members) : Local/queuemem0@ExternalCallFromQueue Local/queuemem1@ExternalCallFromQueue Local/queuemem2@ExternalCallFromQueue

[asterisk-users] Queue handling : does every call have to be accounted twice?

Hi, I have a queue configured on Asterisk (11.14.2); Currently regardless of what codec any side is using, every call handled by the queue is seen as 2 separate calls (4 channels). Calls are incoming to the queue from a (few) SIP trunk(s), queue has the 'queue no answer' option set, autofill is

[asterisk-users] queue periodic-announce without stopping ringing

hello, is it possible to play queue periodic-announce without stopping agents ringing? actual situation is sequential - ring agents, play announce (for 15 sec), ring agents , ... (i need to connect agent with caller asap when agent is free) is it possible with ARI? --

Re: [asterisk-users] queue periodic-announce without stopping ringing

On Mon, Jun 15, 2015 at 9:22 AM, Marek Cervenka cerv...@fpf.slu.cz wrote: hello, is it possible to play queue periodic-announce without stopping agents ringing? actual situation is sequential - ring agents, play announce (for 15 sec), ring agents , ... (i need to connect agent with caller

Re: [asterisk-users] Queue PJSIP, not all contacts rings

Nick Awesome wrote: Hay guys, have question. When I do regular dial I use $this-AGI-get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true); to get all contacts of current endpoint and so I dial to all phones at once, but if I exec QUEUE, I have just one phone rings, seems like

Re: [asterisk-users] Queue PJSIP, not all contacts rings

Works, thank you! On Feb 23, 2015, at 7:11 PM, Joshua Colp jc...@digium.com wrote: Nick Awesome wrote: Hay guys, have question. When I do regular dial I use $this-AGI-get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true); to get all contacts of current endpoint and so I

[asterisk-users] Queue PJSIP, not all contacts rings

Hay guys, have question. When I do regular dial I use $this-AGI-get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true); to get all contacts of current endpoint and so I dial to all phones at once, but if I exec QUEUE, I have just one phone rings, seems like it take first one as

Re: [asterisk-users] queue show queue-name vs queue log for calculating average hold time

On Wed, Jan 28, 2015 at 1:37 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show

Re: [asterisk-users] queue show queue-name vs queue log for calculating average hold time

On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show queue_name I get the following numbers: queue_name has 0 calls (max unlimited) in

[asterisk-users] queue show queue-name vs queue log for calculating average hold time

Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show queue_name I get the following numbers: queue_name has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6%

[asterisk-users] queue reload command

Hi I'm using asterisk 1.8 Does anyone know how to use the queue reload command. The built in help doesn't really help. queue reload {parameters|membe Reload queues, members, queue rules, or parameters Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44

Re: [asterisk-users] queue reload command

reload reception *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ishfaq Malik *Sent:* Thursday, January 8, 2015 2:10 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] queue reload command

Re: [asterisk-users] queue reload command

-Commercial Discussion Subject: [asterisk-users] queue reload command Hi I'm using asterisk 1.8 Does anyone know how to use the queue reload command. The built in help doesn't really help. queue reload {parameters|membe Reload queues, members, queue rules, or parameters Regards Ish

Re: [asterisk-users] queue log realtime mysql

On 04-11-14 11:52, Jonas Kellens wrote: On 04-11-14 11:50, Ishfaq Malik wrote: On 4 November 2014 10:40, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: Hello, I have 5 Asterisk servers all using mysql realtime to store queue log information.

[asterisk-users] queue log realtime mysql

Hello, I have 5 Asterisk servers all using mysql realtime to store queue log information. There is 1 out of 5 servers which stores the data in 4 columns : 'data1' -- 'data 5'. All other servers store data in 1 column 'data' with the data seperated by pipe. I see no difference in my

Re: [asterisk-users] queue log realtime mysql

On 4 November 2014 10:40, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, I have 5 Asterisk servers all using mysql realtime to store queue log information. There is 1 out of 5 servers which stores the data in 4 columns : 'data1' -- 'data 5'. All other servers store data in 1

Re: [asterisk-users] queue log realtime mysql

On 04-11-14 11:50, Ishfaq Malik wrote: On 4 November 2014 10:40, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: Hello, I have 5 Asterisk servers all using mysql realtime to store queue log information. There is 1 out of 5 servers which stores

[asterisk-users] Queue: Passing params to macros and gosubs

I wrote: Queue(ENGLISH,rh,,,20,,,log-answer) but no variables are available in the subroutine context. I need to get ${UNIQUEID} in there. In a Dial command, I can write U(log-answer^${UNIQUEID}) but not in the gosub field of the Queue command. Is it possible to pass variables from the

[asterisk-users] Queue wrapuptime and active calls

Hello, My question is this, I have a service queue that members follow the service interval (wrapuptime = 30). However, sometimes these members need to call the customer back, thus making an active call. Occurs when this member disconnects the call shortly following section in the queue already

[asterisk-users] Queue is not working

Dear All, I have make a queue in my dailplan and queue is not working properly,prbolem is that all call goes to same extenstion at a time.Because,I use eyeBeam(softphone) and eyeBeam have six line and whenever a call comes into eyeBeam that call reserved by Line 1 suppose to 2nd call will come

Re: [asterisk-users] Queue is not working

On 22 May 2014 12:42, omakhileshchand omakhileshch...@gmail.com wrote: Dear All, I have make a queue in my dailplan and queue is not working properly,prbolem is that all call goes to same extenstion at a time.Because,I use eyeBeam(softphone) and eyeBeam have six line and whenever a call

Re: [asterisk-users] Queue is not working

I would research the ringinuse option as well. On 22 May 2014 13:42, omakhileshchand omakhileshch...@gmail.com wrote: Dear All, I have make a queue in my dailplan and queue is not working properly,prbolem is that all call goes to same extenstion at a time.Because,I use eyeBeam(softphone)

Re: [asterisk-users] Queue with linear strategy does not work

On Tue, Dec 10, 2013 at 10:14 PM, Thorben Jensen i...@thorben.dk wrote: I have a queue with linear strategy. When I add dynamic members it does NOT ring the members in the order they are added. I use the command AddQueueMember to add members but it seems to be random how it rings the members.

[asterisk-users] Queue with linear strategy does not work

I have a queue with linear strategy. When I add dynamic members it does NOT ring the members in the order they are added. I use the command AddQueueMember to add members but it seems to be random how it rings the members. Hope somebody can help. This is the description of linear strategy:

[asterisk-users] Queue linear unordered feature when using realtime

Hello, I was trying to use a queue in linear order and to provide the exact order of members to dial by adjusting the uniqueid value. Obviously it doesn't work and it seems an old problem: https://issues.asterisk.org/jira/browse/ASTERISK-18480 Realtime configuration can't identify orders in the

Re: [asterisk-users] Queue linear unordered feature when using realtime

From: Leandro Dardini [mailto:ldard...@gmail.com] Sent: Thursday, November 14, 2013 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Queue linear unordered feature when using realtime Hello, I was trying to use a queue in linear order and to provide

Re: [asterisk-users] Queue Management

This should happen automatically - not sure what you want to do. l. 2013/9/26 akhilesh chand omakhileshch...@gmail.com: Dear All, I have six different campaign and 5 different agent have login on that campaign.Same thing i have done using agi and database,i never use queue management on

[asterisk-users] Queue Management

Dear All, I have six different campaign and 5 different agent have login on that campaign.*Same thing i have done using agi and database,i never use queue management on this scenario. Agent** can also shuffling one campaign to anther campaign. * Now i want to do some work with queue.I want to

Re: [asterisk-users] queue member ackcall - cpuspikes

And further, No matter what contains in the GOSUB (In this case relatively simple stuff), when the A party hangup, the queue should signal the B Channel(Member) to hangup. [ Which should tear down member LEG immediately ] The problem here is Queue is not able to hangup the member leg even though

Re: [asterisk-users] queue member ackcall - cpuspikes

hi! GOSUB X * Presents Background message to the called party * check if there's any inputs from the user ( Press 1 etc ) * exit if called party provide input *or not* See the example URL for for similar implementations. Regds On Thu, Aug 8, 2013 at 2:03 PM, Paul Belanger

[asterisk-users] queue member ackcall - cpuspikes

hi!, Asterisk Version:1.6.1.20 OS: CentOS release 5.3 (Final) uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386 GNU/Linux Application: Queue Specific Details: Obtain Acknowledgement from queue member before bridging the caller. Language: AEL Similar

Re: [asterisk-users] queue member ackcall - cpuspikes

On 13-08-07 08:42 PM, zendel fernandez wrote: hi!, Asterisk Version:1.6.1.20 OS: CentOS release 5.3 (Final) uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386 GNU/Linux Application: Queue Specific Details: Obtain Acknowledgement from queue member before bridging the

[asterisk-users] queue stats last reset date/time

Hello everybody, is there any way to find out when the queue stats ('queue show' / AMI action 'QueueStatus') was last reset (by 'queue reset stats')? These counters would make much more sense if I knew what timeframe they cover. ;) -- marie --

Re: [asterisk-users] Queue - how to jump to next member after NO ANSWER?

, July 23, 2013 3:04 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Queue - how to jump to next member after NO ANSWER? ** ** Hi all, I have a Queue with 3 members: SIP/100 SIP/200 SIP/300 When call arrives SIP/100

[asterisk-users] Queue - how to jump to next member after NO ANSWER?

Hi all, I have a Queue with 3 members: SIP/100 SIP/200 SIP/300 When call arrives SIP/100 is ringing.. After given timeout ringing stops but call is not routed to next member but SIP/100 starts ringing again. I know that this is because SIP/100 is still available in the Queue but is it any way

Re: [asterisk-users] Queue - how to jump to next member after NO ANSWER?

: [asterisk-users] Queue - how to jump to next member after NO ANSWER? Hi all, I have a Queue with 3 members: SIP/100 SIP/200 SIP/300 When call arrives SIP/100 is ringing.. After given timeout ringing stops but call is not routed to next member but SIP/100 starts ringing again. I know

Re: [asterisk-users] queue moh

Have you looked at mohsuggest in the sip configuration? Regards Ish On 10 July 2013 17:55, Andrew Thomas a...@datavox.co.uk wrote: Hi All, Sorry if this has been covered already, but I don't tend to follow this list as close as I should these days. Problem is that if a call comes in to

[asterisk-users] queue moh

Hi All, Sorry if this has been covered already, but I don't tend to follow this list as close as I should these days. Problem is that if a call comes in to a queue without option 'r' specified - moh plays as expected. Now, when that call is answered, all is fine. Trouble comes when that person

Re: [asterisk-users] queue moh

Hello Andy, Have you tried using SetMusicOnHold command before Queue command? BR, Ioan On Wed, Jul 10, 2013 at 7:55 PM, Andrew Thomas a...@datavox.co.uk wrote: Hi All, Sorry if this has been covered already, but I don't tend to follow this list as close as I should these days. Problem

Re: [asterisk-users] Queue questions - Asterisk 11

Le 04/07/2013 07:29, Satish Barot a écrit : [...] Already tested, I tried again as the option passed to queue was changed (n option) Logs: -- Started music on hold, class 'default', on SIP/gw-005e -- Executing [909@memberconnector:1] Dial(Local/909@

Re: [asterisk-users] Queue questions - Asterisk 11

On Thu, Jul 4, 2013 at 5:36 PM, Administrator TOOTAI ad...@tootai.netwrote: Le 04/07/2013 07:29, Satish Barot a écrit : [...] Already tested, I tried again as the option passed to queue was changed (n option) Logs: -- Started music on hold, class 'default', on

Re: [asterisk-users] Queue questions - Asterisk 11

On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI ad...@tootai.netwrote: Hi all, I have to questions about queues. Member is a phone like SIP/myphone and only one member in the queue. At first, DIALSTATUS doesn't return any status. How to now if a call in queue has been answered or if

Re: [asterisk-users] Queue questions - Asterisk 11

Hi Satish Le 03/07/2013 09:15, Satish Barot a écrit : On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: Hi all, I have to questions about queues. Member is a phone like SIP/myphone and only one member in the queue. At

Re: [asterisk-users] Queue questions - Asterisk 11

On Wed, Jul 3, 2013 at 2:37 PM, Administrator TOOTAI ad...@tootai.netwrote: Hi Satish Le 03/07/2013 09:15, Satish Barot a écrit : On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI ad...@tootai.netmailto: ad...@tootai.net wrote: Hi all, I have to questions about queues.

Re: [asterisk-users] Queue questions - Asterisk 11

Le 03/07/2013 15:07, Satish Barot a écrit : [...] Then you should add Local channel as a queue member and dial your SIP member from Local channel context. A little hint here. Suppose you have a support queue configured in queues.conf ;queues.conf [support] ... ... member =

Re: [asterisk-users] Queue questions - Asterisk 11

On Wed, Jul 3, 2013 at 7:40 PM, Administrator TOOTAI ad...@tootai.netwrote: Le 03/07/2013 15:07, Satish Barot a écrit : [...] Then you should add Local channel as a queue member and dial your SIP member from Local channel context. A little hint here. Suppose you have a support queue

[asterisk-users] Queue questions - Asterisk 11

Hi all, I have to questions about queues. Member is a phone like SIP/myphone and only one member in the queue. At first, DIALSTATUS doesn't return any status. How to now if a call in queue has been answered or if caller just hangup? Second, how to deal with timeout, I have strange

Re: [asterisk-users] Queue Ring inuse is shared ?

I have 1.8.7.0, Realtime queue table with ringinuse set to 0, callcounter set to yes in sip .conf for my SIP members. Above allows me Queue not sending a call to a member when (s)he is on call(Be it from same Queue or any other call). Member can also transfer(through features.conf) a call without

Re: [asterisk-users] Queue Ring inuse is shared ?

On 22 June 2013 10:11, Shanavaz E A shanava...@yahoo.com wrote: Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. Should that not be ringinuse = no? -Barry --

Re: [asterisk-users] Queue Ring inuse is shared ?

. From: Shanavaz E A shanava...@yahoo.com To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Saturday, June 22, 2013 1:11 PM Subject: [asterisk-users] Queue Ring inuse is shared ? Hi, I use asterisk 1.8. My issue is : I have the same SIP

[asterisk-users] Queue Ring inuse is shared ?

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension

Re: [asterisk-users] Queue Limit Callers

Subject: Re: [asterisk-users] Queue Limit Callers Hello Shanavaz., Please find some quick thoughts: * 2 main queues * agents logged on one or on both main queues * before sending a new call to one of the main queues check the number of waiting callers (QUEUE_WAITING_COUNT function

Re: [asterisk-users] Queue Limit Callers

On 17 June 2013 11:02, Shanavaz E A shanava...@yahoo.com wrote: Hi, I have a requirement, which I am not sure whether it can be implemented. I had done some searches but didnt find an answer to this. Kindly let me know if some one has an idea to implement this: I am not aware of an

Re: [asterisk-users] Queue Limit Callers

You should have different sets of agents logged in to different queues and you should have a monitor to move them from one queue to the other based on incoming traffic. l. 2013/6/17 Shanavaz E A shanava...@yahoo.com Hi, I have a requirement, which I am not sure whether it can be implemented.

Re: [asterisk-users] Queue Limit Callers

Hello Shanavaz., Please find some quick thoughts: * 2 main queues * agents logged on one or on both main queues * before sending a new call to one of the main queues check the number of waiting callers (QUEUE_WAITING_COUNT function) and divert (for example for 30 sec) the call on a empty members

[asterisk-users] Queue Limit Callers

Hi, I have a requirement, which I am not sure whether it can be implemented. I had done some searches but didnt find an answer to this. Kindly let me know if some one has an idea to implement this: I have two Queues - Sales Booking I have 12 Agents who are added to both the queues Suppose

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