similar to: To queue or not to queue...

Hello; If I have load up to 220 extensions with 50 concurrent calls. Can one hardware server carry all this load? What is the hardware server required for this? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130714/b63756d2/attachment.htm>

Hey All, Growing call center. Currently at about 200 call center staff, running about 1000 calls per hour. Gearing up to double that. Not too sure that a single server will support that growth. So, I'm trying to come up with ways to scale the system and still maintain a simplistic design. So I'd like to bounce some ideas around. Currently I am running on a Dell 1950, dual quad core

Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? Thanks, Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090408/f6d5af5a/attachment.htm

Matt, At some point you need to consider how much is too much... I run a call center with more then 125 commissioned phone sales reps and more than 60 customer service reps. We run dual servers, fiber from one provider and 6 bonded T1's from another provider. We purchase our so trunks from a wholesale company who is a major provider to resellers. Being so, their network is extremely

Hello All, I have a need to connect an analog device to an asterisk server. The analog device has 4 analog lines going into it (it?s a fax solution). The fax solution answers the analog call, then listens for dtmf. The dtmf code that is played tells the fax device what email address to send the fax to. All calls on our system come into the server through a PRI. The faxes come in over a PRI,

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Say I have agents using a softphone like eyebeam that has 6 lines. They log in to the queue. Say there are 3 agents in my queue. 3 calls come in and all three agents are on a call. Now a fourth call comes in. Is it possible to have it setup so that the 4 call rings on line 2 of one of my agents, if they don't get it within the time limit it rings on line 2 of another agent and so on. An

Queue not sending call to Agent I am having an issue and i am not sure if it is a bug or a config issue. I was originally running Asterisk 1.8.1.1 when I noticed this issue. I upgraded to 1.8.4.2 to see if that would fix it but it didn't. The issue is that I have a call queue and the agent dials a number to log into the queue. When someone calls the queue the first time the call is

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 use single queue to managing this. Eg: campaign Agent Login A a_1,a_3

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

Can anyone tell me what might cause callers on hold in a queue to drop into agents voicemail boxes? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071217/ae6cba5a/attachment.htm

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 there are 12 calls in the Booking Queue, and 6 calls in the Sales Queue. Only 8 calls in the

Trying to figure out the best way to pull an active call out of a queue by unique id and put it on hold. I don't want to put it on hold on the agent's phone but I want it to be pulled away from the agent's phone and into Asterisk limbo somewhere. Shortly after I want to pull the same call out of limbo and redirect it back to either the same agent or another. I was thinking about call

We're moving a somewhat complicated call center over to an Asterisk system, and I'm looking for documentation on queue/agent configuration. So far I haven't found anything on the Digium or Asterisk websites, and I was hoping that someone could point me in the right direction. Thanks, Devon

Hi, I've been configuring AsteriskNOW from the GUI but could it be that the GUI isn't working properly? because when I make a queue and add a few agents, and when I call the queue none of the phones ring. The queue is also configured at "Ringall" I checked the queues.conf file and the settings matched. I also noticed that the agents I made in the GUI, that they were not written

Hi all, I am trying to write an application to monitor queues using the Asterisk Management API. So far I have had some level of sucess, basically reverse engineering the protocol and the event messages using ethereal etc. I know there are a couple of pages on the Wiki that attempt (no dis-respect to who ever did it as it has been a great help) to document the API and was wondering if there

Hei. I have a question about how to get the periodic-announce to work within my queues. I got the following test: extensions.conf --> exten => s,2,Queue(test|rtT|||200) Queues.conf --> [testqueues] strategy = ringall context = testcontext timeout = 250 periodic-announce-frequency=60 periodic-announce = queue-periodic-announce member => SIP/591 Log from " show queues "

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 that call goes to Line 2(same extension used by Line 1) and 3rd call goes to 3rd line and so on. But i

This is the problem: Call coming into a queue in ringall strategy, if a member (SIP) of the queue is busy when entering the queue, and this member comes free after a little time, the member never rings.. How to solve this? I changed all parameters of the queue with no results... Wath i need: If one member of the queue is busy when a new call come in to the queue, this member can hangup and

We have a queue monitoring application running so we can see the caller ID of callers in a queue. If we see a VIP in the queue, is there any method to force that call to be first in line? If there's a softphone, or queue managing application already written that does this, I'd love to know.