similar to: Local agent/member in-use after transfer

Hello everybody, we are using asterisk 16 with a realtime config and have a problem with FOP2. We have developed a webinterface for managing the queues. If we add a member to a queue, everything works fine but the user is not shown in the queue in FOP2 Panel. The problem is that the FOP2 Panel does not receive the QueueMemberAdded Event. This will only be sent if the QueueAdd Function is

Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within

When agents are on the phone, and the CLI queue show command shows their status as busy, the queue still tries to send them calls. Running Asterisk 1.6.0.17 and using AddQueueMember to dynamically add agents. ringinuse is set to no for queue. Agents are using Polycom 430s. dialplan: exten => s,n,Queue(orders,itT,,,80) queue definition in queues.conf: [orders] maxlen=20 queue-thankyou=

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt

Hi... I just had a real quick and simple question... I have a asterisk implementation setup w/ real time off of a mySQL database for SIP peers and queues, voicemail, agents etc... I after the upgrade to asterisk 1.4.3 there are some new configuration features i would like to use. I was wondering if i could just add to the database table a column for the new config option? if this will work or

My settings for the queue.log are in the [general] section of logger.conf I'm running 13, I didn't see what version you said you were running. If I wanted to add a LOCAL channel to my queue I'd do it as member => LOCAL/7124 at kiniston-intern,0,John,hint:7124 at kiniston-intern On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch <idemkovitch at yahoo.com> wrote: > John,

Hi , I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 & 101 ) in a queue..When a caller arrives in queue , it lands on first 100 , 100 then does a blind transfer to 101 .. so that the caller can converse with 101 .. strangely enough the queue_log shows : 1253814090|1253814090.12|55365|NONE|ENTERQUEUE||98221232123

Just a silly question that I'm not sure. Ringinuse is working with IAX in 1.6??? like in sip?? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090306/fa56ebfc/attachment.htm

Hi all, I have agents who are members of more than one queue. When an agent is busy with queue A, he is not considered busy by queue B, and receives call (since his EyeBeam Softphone has 6 channels). Besides that, I use a monitoring tool that connects through the manager interfaces and run "show queues" and "show agents" to know agents statuses. I need Asterisk to consider

Hi, I am using asterisk-1.4.15, My sip configs is like [2501] type=friend username=2501 secret=2501 canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip disallow=all allow=ulaw incominglimit=1 nat=1 queue.conf is like [gen-enq] joinempty = yes musiconhold = default strategy = rrmemory servicelevel = 60 timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes

I posted this over in asterisk-dev, realized I probably should have put it here. Hi there, We?ve been having a strange issue with a customer?s queues where a queued call will ring an available agent, agent answers, then a second or two later the agent is offered a second call which they cannot answer, since they?re already speaking with a client. This in turn causes a few issues: - Agent stats

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 is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

Hi, I'm sure some else has been facing this problem. I want to record all the call coming in my queue. I want this format: YYYYMMDD-HHMMSS-AgentID-CallerId - UniqueID. I'm using the monitor feature inside the queue.conf. I can't use the agents.conf monitor features because I'm using dynamic agent (addqueuemember) The problem I'm facing is that I can

I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses

Good Day list, Anyone know if there is a way to have the AgentCallBackLogin function play a voice file after the agent picks up the phone? If this is not an available feature, any ideas on the difficulty in making this feature? Example: Extensions.conf exten?=>?700,1,AgentCallbackLogin(${CALLERIDNUM}|?AnnounceCAllQue-TechSu pport?); ....... exten => s,6,Queue(queue1)

Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin.

Hello everyone. I am using ggplot and I need some help to merge these two plots into one. plot_CR<-function(x,y,agentid,CRagent){   library(ggplot2)     agent<-CRagent[[agentid]] # To make following expression shorter   ggplot((data.frame(x=CRX,y=CRY,sr=agent$sr)))+   geom_point(aes(x,y,colour=cut(sr,c(0,-10,-20,-30,-40,-50,-60,-70,-80))))+   geom_text(aes(x,y,color=cut(sr,

Hi all! My problem is that calls being placed in the queue, and are waiting while the agents are busy, when an agents is then free they gets connected to the agent but there is silence (no voice). If a caller has not to wait in the queue, there is no problem. My agents have an iax2 client, and imcoming calls are over SIP. queue.conf: persistentmembers=yes autofill=yes ringinuse=no