similar to: Round Robin Call Distribution

Hi, I have several agents, who all log into multiple queues. What I want to happen (but doesn't seem to be) is: Agent 5 is logged into queues 1,2,3 Agent 4 is logged into queues 1,3 A call comes into queue 1, and goes to agent 5. Agent 5 answers the call and finishes it. A call comes into queue 3. I want this call to go to Agent 4, as opposed to going to agent 5 (which is what it is doing

Hi all, I am seeing a strange problem with Asterisk queue. I am not sure if it's my configuration which is wrong or there's something with Asterisk. I am using Asterisk 1.4.2 and i have a queue with one MGCP member. When i tried to call the extension number directing to the queue, the MGCP phone is not ringing. However, it is fine to call the MGCP phone directly. The strange thing is

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hey guys, i'm wondering if there is any good way to get app_queue working in real roundrobin strategy. The idea is to specify a call list of, lets say, 3 agants. Those agents should always be called in the correct defined order. So all calls have to get the following agent priority: 1st Agent -> 2nd Agent -> 3rd Agent I've actually solved that by defining penelty for the accounts,

I've noticed that when app_queue.so is reloaded(or just a reload command is used) that all queue members that were paused are automatically unpaused. Is there a workaround for this? (Note, I use statically defined callback agents). --johann

Hi All, I know I can do this pretty easily with one of the SIP Proxy/Routers, I already do this using OpenSER as a load balancer. I have a special requirement that insist an Asterisk server, 1.6.1.x, is used. I will have 2 SIP trunks coming into the server and I will have to send calls to these SIP trunks with a round robin distribution pattern. I was thinking of using a group count

Hi, I'm a little bit stuck with Queue app. I'm putting callers into the queue and have them hear music on hold when all (static) agents are busy. This is easy. But when agent become available I want the caller to hear a ringing tone (with message that his call has been routed to the support representative). Is this somehow doable? Thanks, David

Does anyone know what can cause queue members to go into a status of "Unknown"? pbxtel-01*CLI> queue show cs has 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime), W:0, C:447, A:20, SL:91.7% within 60s Members: SIP/1405 (dynamic) (Unknown) has taken no calls yet SIP/1420 (dynamic) (paused) (Not in use) has taken no calls yet SIP/1442

When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device

Hi, I'm writing a Java client/server application that talks to the Asterisk manager interface via the asterisk-java stuff. The idea being it will give you an app to run on your desktop that monitors your phone essentially. Once I've got something vaguely working it will be released under the GPL and hopefully people will contribute to it etc... As part of this, I'm currently

Hi all, I am having the following trouble with recording calls: When calls come into the support line did number, the call starts to record on the first queue, but appears to hang up when the call actually connects to the engineer (ie I see "got hangup request" on the cli and then mixmonitor ends.) I am guessing this has to do with the announce file that is played to the engineer

Hi, I dont really know if this is the right place to post this question..If its not pl let me know... Here is what i am looking for. I have two machines with two ethernet cards.One of these machine has an ftp server(vsftpd). When i request a file from the first machine i want that machine be able to start an ftp, and when each of the packet of the file go to one of the ethernet cards, i want

I was wondering if anyone has an easy way to emulate dialing in a round robin fashion like when you use Zap/r1 for Zap trunks. At the moment what I do is simply make a macro that will dial the sip trunks in order so if the first one fails it goes to the second and so on. The problem with this approach is that the first few SIP trunks will always be busy because of outgoing traffic. Is there an

I recollect that there used to be a fixed, finite limit to the number of call groups that could exist. Does anyone know if that limitation still exists in 1.2.1, or maybe where I could look in the code to find out if it's a fixed length array or not? Thanks. Doug.

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia

I'm pretty sure I already know the answer to this, but... Is there a way to copy/transfer/replicate sip subscriptions from one asterisk system to another, for the purposes of HA? You coudln't even write a script to do it I don't think. You can do an 'asterisk -rx sip show subscriptions' but there'd be no way to repopulate it on a second system. Yes/No? Doug.

I need to set up an office full of Cisco 7960 phones behind NAT with the server out in Colo. The first test phone registers fine, but the second one does not register. The first phone's registration looks like so: /SIP/Registry/3115552368 :64.169.xx.yyy:38836:3600:3115552368:sip:3115552368@64.169.xx.yyy:5060 When the second phone tries to register, it gets back a 404 not found. Not a

> > The internal DNS is NOT supporting round robin. As Rowland said use Bind9 > That's news to me! If so, then the internal DNS backed is not suitable for multiple DC's. (Though, I could've sworn it worked on versions 4.2+ < 4.7. It's on my to-do list to explore this further with different versions.)

Hi, I have several Polycom 501 connected to asterisk. The phone has an ACD-login function that I'd like to use. But I can't find find much information about this. I've read a post on bugs@digium (http://bugs.digium.com/view.php?id=6119) about this function but I'm not really clear on if this is actually working or not? Has anyone actually used the Polycom ACD-login function

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com