similar to: How to configure Asterisk queue with Vonage account?

Followed; http://www.voip-info.org/wiki-Asterisk+Agents agents.conf [agents] agent => 1001,4321,Ben Dover queues.conf [queue1] member => Agent/1001 extensions.conf exten => 28,1,AgentLogin(1001) exten => 29,1,Queue(queue1) But when I call number 28, I get: "Please enter your password followed by the pound key".. but when I enter the the password, 4321,

Hi, I'm playing with the agent/queue system. Everything work well with v1.0.3. but I want the 'Action: Agents' in the manager API that is only on the CVS version. So i switched to, but now the Queue/Agent system barely work. (my agent don't get the call) Where I can get a 'stable' CVS version? Or maybe, how I can solve my Queue/Problem? here is the detail: 1. I can

Hi, I'm having hard time to emulate agencallbacklogin. Agent can logon and receive call without any problem using addqueuemember. The problem comes when I try to evaluate their performance using queuemetrics. Here is an exemple of my log script: ;Agent Login exten => _60XXX,1,Macro(agentLogin) [macro-agentlogin] exten => standard,1,AddQueueMember(queue1) exten =>

Tzafrir Cohen wrote: > > The agent picks up the phone but neither the agent nor the caller > > > here anything. >So please provide a more complte trace and a the relevant partt of your >dialplan. > Here is the relevant part of the dialplan: [local] exten => 98,1,Dial(SIP/sguenther,20,tr) exten => 98,2,VoiceMail(98|u) exten => 98,3,hangup exten =>

How to get rid on following.. why its Invalid ? holler*CLI> queue show queue1 queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet No Callers -------------- next part -------------- An HTML

Hi, I am running a call center based on Asterisk and building some statistics based on the queue_log file. I have some doubts about it that maybe you could help (actually, maybe these doubts are suggestions for enhancements!): 1st Scenario - Agent receives the call, and puts it on parking for somebody else to pick it up. Parking # 7000 (for attender transfer)

We need to make sure that all queue applications run on the correct system that the user agents that own the queue application are registered to. So when a server fails and the user agents register with their secondary server (which will always be configured to be the same server for those related agents) the queue application is running on that server and routed to correctly by it's peers.

Paul wrote > >;Pause/unpause Queue >exten => 424,1,PauseQueueMember(|SIP/${CALLERID(num)}) >exten => 424,2,Playback(unavailable) >exten => 424,3,Hangup >exten => 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)}) >exten => 425,2,Playback(available) >exten => 425,3,Hangup > Following your suggestion and a number of postings and articles I have

As I understand it, you'd enter the extension at which you wish to be called back at, your 9665 has nothing to do with it. Instead of dialling 28 you could dial 9665 and that would add that SIP phone as an agent to the cytelcs queue. Steve -----Original Message----- From: Harold Workman [mailto:hworkman@cytelcom.com] Sent: 22 June 2004 18:54 To: asterisk-users@lists.digium.com Subject:

Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 <- call1 queue1 <- call2 queue2 <- call3 queue2 <- call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could

Hi, I''m trying to use HTB for IPv6 packets. In the LARTC archive I found a thread on this issue where a solution was described. The idea is to mark a user-defined chain then use the fwmark for filtering. However my attempt wasn''t successful. I''m thinking of trying my luck with the u32 filter. This is what I did, any comment is greatly appreciated. Thank you all. tc

I have a centrally located shared imap folder location. We have several maildir folders with different file system permissions. They are for several groups of people who are loosely related for support purposes. All mail gets delivered to queue1 just fine. The files are group writable by group1. Some of the mail gets moved via imap to queue2. Queue2's maildir tree is configured with

Hmm i found the problem... I?m using a Grandstream BT100. The transfer just works in a queue if I first acknowledged the call using the # key, and then press the TRANSFER key in the Grandstream. In the asterisk console I receive a: -- SIP/4002-4563 acknowledged Then I can transfer the call... Weird because i?m using ackcall=NO in agents.conf ... Diego Magalh?es diego@redetaho.com.br +55 24

Which kind of transfer do you use? Try using the # transfer. Hope that helps.. Guido Hecken -----Urspr?ngliche Nachricht----- Von: Diego Magalh?es [mailto:diego@redetaho.com.br] Gesendet: Mittwoch, 2. Februar 2005 17:21 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] AgentLogin / AgentCallbackLogin transfer problem Hello guys, I?m running Asterisk CVS-HEAD-02/01/05-12:22:46 and

Hello guys, I?m running Asterisk CVS-HEAD-02/01/05-12:22:46 and having a problem with call transfers using the cmds AgentCallBackLogin and AgentLogin First Case (using cmd AgentCallbacklogin): When the incoming call comes and enters the queue, the agent logged in answer the call. But when I try to transfer this call to another agent, the incoming call is dropped. I don?t receive any error

Is it possible to distribute QUEUE information among several Asterisk nodes in a "multimaster" or "load balancing" setup? I haven't tried this yet but if one uses realtime with a clustered multimaster database and the queue agents/members are fixed SIP channels (eg. SIP/100) then I guess that the Queue app will be able to contact the member no matter to which Asterisk node

> -----Original Message----- > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > Sent: Thursday, June 15, 2006 10:36 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] DUNDi Not Able to > HandleComplexFailoverSituations > > > Is it possible for you to explain in more detail the > situation involved.

I setup a vonage account last year, and cancelled it last night when I put my asterisk box together and signed up for a Broadvoice account to use with it. Now I would like to use my Linksys router as an MTA, but realize it is still programmed with all of vonage's proprietary information and I do not know how to clear it. I understand that just pushing the reset button will not do it.

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

Does anyone currently use Vonage with Asterisk? I've tried to set it up but it looks like Asterisk (at least the version that I have) does not handle well the SIP call dialog, sending a BYE with the wrong tag. As a result, when I hang up, Vonage sends back a 400 Bad Request and the call on the PSTN side does not hang up. I know that Vonage does a lot of nasty stuff which impacts UA's