similar to: Re: Asterisk-Users Digest, Vol 13, Issue 131

Hi, For some reason, I seem to have issues with dailing toll free numbers and can't seem to find out why, sometimes, I get a busy signal. Some other times I get weird errors from the phone. The error below was a simple busy signal. Here's couple of my info relevant to the problem: -- Reconfigured channel 1, PRI Signalling signalling -- Reconfigured channel 2, PRI Signalling

My first public-facing deployment of Puppet is likely to be on some parallel computing nodes we have here. One of our fluid dynamics packages expects to be able to rsh from one master node to other slave nodes in the same batch queue. I think I''ve made some good class definitions up to this last point, where code duplication rears its head. Example of the problem: class ch405-host

hi list, i have a question about the args of queue: when we use Queue() app, there are some arguments than can use. help from CLI: Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule]]]]]]]]) well.. i'm trying to identify who has taken the call on a queue, and, when agent conected, launch a macro with some args based on who takes the call i do: exten =>

Dear all I've got call queuing working when calls originate from my local site. After testing I migrated it to calls originating from our voip provider- it should ring an extension, then queue . All works well apart from if the caller hangs up when queued: the call hangs around in the queue as a phantom until one of the extensions answers it and it is destroyed Am I doing something wrong?

----- Original Message ----- > From: "John Kiniston" <johnkiniston at gmail.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Tuesday, July 28, 2015 12:12:05 PM > Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are registered > > In your queues.conf do you

Hi List, Just one more question that may sounds stupid to some people but I can't find the solution for now, I have the following dialplan: exten => queue,n,Queue(myqueue) exten => queue,n,NoOp(ENDQUEUE) I don't understand why the NoOp is never triggered, the incoming call is always hangup when the agent hangup... Is this a behaviour we can't get rid off without patching

Hello, I am running Asterisk 11 on CentOS 6.x. I have configured several queues as follows in extensions.conf: exten => s,1,Queue(myqueue,rtnC,18) same => n,Background(user_unavail) same => n,WaitExten(10) exten => 1,1,Voicemail(1111 at my-vm,s) This rings the phones in the queue for 18 seconds. If no queue members answer, the caller is then prompted to press 1 and leave a

Hello list, when I sent an incoming call first to a queue and after the timeout to a dial-command, while the correspondent's phone rings there is no ringtone for the caller... So it goes like this : 1. dial(SIP/account1,20) 2. queue(myqueue,,,,20) 3. dial(SIP/account2) In step 1 there is a ringtone for the caller. In step 2 there is musiconhold (class default) for the caller. In step 3

Hello list, I am trying to set a custom SIP header on all calls that are made by the app queue because I want to track a certain state at the SIP level. If I use the following code: exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID}) exten => s,n,Queue(myQueue) this works fine for the FIRST call made from the queue to an agent; but if that call does not go through, it's not repeated

I've got another issue I can't quite figure out and a search of the archives and Google turn up nothing... Say a call comes in (these are all via SIP) and is sent into a Queue(myqueue,t,,,300). Note the "t" to allow whomever receives the call to transfer it. The call is enqueued, and the logged in agents ring as expected. An agent answers the call, and begins chatting. He

Hi I am facing an issue where asterisk cannot see the queues that exist in my database through realtime. I am using res_odbc and a local mysql database. If I run: realtime load queues name myqueue I get "No rows found matching search criteria.", however if I do the same for a peer: realtime load sippeers name Then I get a result. Since my queues table is in the same database as my

Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue "myqueue" In extension file: Queue(myqueue|t|||120) And my agents are joining in following

Hi, I have the same problem with the queue configuration When I receive 2 calls only 1 phone ring even if more agent's phone are free. The second call will go to an other agent only if the first call is pickup. Somebody have a solution ? This is my config file : Queue.conf [general] ; ; Global settings for call queues ; ; Persistent Members ; Store each dynamic agent in each queue

Hi!, I'm trying to set up a Queue (which works fine now :-) Sip clients can login in to the Queue with dialing 91 on there phone. And as soon as there are customers the Queue calls the agents back. I would like that the queue calls the agents also if it's phone is call-forwarded. With agents (sip clients) are added with the following extensions: exten => 91,1,AddQueueMember(myqueue)

Hello ALL! 1. eviroment: 1) PBX:Asterisk@HOME(Asterisk 1.2.7.1) 2)Media GateWay:Grandstream HT 488(register to pbx using SIP account,such as 9001),connect to PSTN(phone no is 82820088) using its FXO. 3)Queue:set up a queue named [myqueue],dial 2020 can call queue 2.my question: when customers need help ,they call 82820088, and then call 2020 to enter the queue . From now on ,the

I have asterisk 1.4 using Queue application. I have this error I must restart Asterisk to correct it. Any Ideas?? log: [Jun 20 16:42:27] WARNING[29339] channel.c: Unexpected control subclass '17' [Jun 20 16:42:38] NOTICE[29337] app_queue.c: No one is answering queue 'myqueue' (13/12/0) [Jun 20 16:44:16] WARNING[30203] file.c: Failed to write frame [Jun 20 16:45:11] WARNING[8044]

Hi, On a 11.8.1 system, I'm trying to configure a queue in which busy agents do no get incoming calls. I dynamically add agents with lines (shamelessly copied from a running Freepbx/Asterisk 1.8 system) such as: AddQueueMember(myqueue,Local/123 at agent/n,10,,hint:123 at subs) Basically, it works but incoming calls are sent to busy agents. In sip.conf, callcounter is set to yes. With CLI,

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

Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it

Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one