similar to: Qeuee/Agent Question

Hi list, Using Asterisk 1.2.10 I am getting seriously confused by Queues and Agents. So far I configured my queue and agents, had my agents login using agentcallback. Call enters queue agent gets a call, if agent doesn't answer after 20 seconds a flag is set in AstDB (thanks to: Leo Ann Boon), call is returned to queue and the cycle continues. If the same agent doesn't

Greetings to all. I am currently implementing call queues for a customer and have come across several "problems". The customer is an airline representative, and will be using call queues for different airline reservations. The customer requires that any agent be able to login to any number of queues. This means that queue members have to be dynamic, not using "member =>

Hello. I've got 1 queue setup with 2 possible agents. Agent 1 is logged in and awaiting a call via AgentCallback. Agent 2 has not logged in. An outsider (caller A) calls in and is placed in the queue, cytelcs. Agent 1's phone rings and Agent1 and A talk. While they are talking, caller B calls in. Caller B is correctly placed in the queue and hears music, however this shows up in asterisk

Hi all, This is to let you know that I found out how to automate the agentcallback logon and logoff. Only thing you need, is to have the agentcode and pincode available in channelvariables. I've updated the documentation on voip-info to incorporate my findings. http://www.voip-info.org/wiki-Asterisk+cmd+AgentCallbackLogin Have fun with it! Regards, Michel Koenen

Hi List, I'm attempting to set up a queue and agents using agent call back. This is all working fine with the queue and the agents login etc However. In my dial plan I a set variable when a call is entered into the queue to identify the origin of the call, then when the agent is called I test to see if the call is from the queue. If it is, the dial plan does not go to VM if the

hi, all i try to confiture realtime queue, but not work, details as below: Insert into queue_table(name)value('95040654321'); INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun', '95040654321', 'SIP/1001', 2, 1); INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321', 'SIP/1002', 2, 1); INSERT INTO

I am trying to find the mean for the elements in the vector Incubation=as.POSIXlt(OnsetTime)-as.POSIXlt(MealTime) where OnsetTime=c(NA,"1940-04-19 00:30","1940-04-19 00:30","1940-04-19 00:30",NA,"1940-04-18 22:30","1940-04-18 22:30","1940-04-19 02:00","1940-04-19 01:00","1940-04-18

The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing

Hello, I have a partially working Realtime Queue using unixODBC and Firebird. I followed the steps on asterisk-guru and the voip-info asterisk message boards but have been having no success. I say that my queue is partially working because callers can join the queue. However the calls do not go to the agents even if the agent is not tied up and is available for calls. So the caller is stuck

Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks

Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general]

Hi, I run a small inbound callcenter with 3 agents doing techsupport. The agents are logged in via softphone, using agentcallback login. Some times the agents PC running softphone hangs, and they reboot the PC. But * is not aware of this and tries to send calls to the PC, which gets rejected. -- outgoing agentcall, to agent '1009', on 'Local/1002@from-sip-c3fa,1' --

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

I have several remote phones that experience a slight "call" delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? Any idea what could be causing this? Thanks, Bill. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all, I try to plot a logstic regression model as follows: Valkror<- read.table("Valkror.txt", header=T) np.bark<- glm(Npinc~bark, family=binomial, data=Valkror) plot.formula(np.bark$fitted.values~ Valkror$bark) The above looks nice but I want to connect the points with a line so I try: plot.formula(np.bark$fitted.values~ Valkror$bark, type="b") but it gives a

I am trying to configure a new server for use in a small Call Center. I want to use realtime queues and agents and after following the instructions I can get the queue to show up on the system but no agents. I am using Asterisk 1.4.4 on a CentOS 5 machine. I have this in extconfig.conf: queues => mysql,asteriskcdrdb,queue_table queue_members => mysql,asteriskcdrdb,queue_member_table I put

The default verb for match is :any. This means for: match ''dog/bark(/:num)'' => ''dogs#bark'' # get 3 barks match ''dog/eat'' => ''dogs#eat'' # post breakfast someone could post on dog/bark with :louder or get on dog/eat with :breakfast. This is annoying and messy, and definitely not RESTful. In moving to Rails 3,

I have asked a GSM operator in my country if he can route a number or a short code to my asterisk server via SIP (since they dont give DIDs in my country) the operator said they do not support SIP, they have no way of converting GSM calls to SIP to then send them to me. I would like to know what is needed from the operator side to do this, what kind of material is needed, or what can be done from

I (have to) would like to move my agents out of agents.conf in preparation for the deprecation of agentcallback login. Everyone I have spoken to is upset about this but the functionality can be accomplished in the dialplan and that is fine by me. I do have an issue with losing the features contained in the agents.conf though. I have to have the ackcall-yes working. All of my agents login on home

Hi Jean-Marc, As I told you, bank is a reserved keyword in Analog Devices compiler for Blackfin architecture. So we need to change the variables named bank to something else. Here's a patch that changes bank to bnk in the 3 concerned files. (Hope the format is OK) About my previous problems with the Blackfin: -> strange block repetition that could be cancelled by the AEC I was busy