similar to: queues and LOCAL for members

Hi, I have an simple queue and agents defines with memeber => SIP/123. If for example Agent "SIP/123" has an call, the queue didnt care and tries to send additional calls to this agents. So Iam loosing time. SIP/123 (In use) has taken no calls yet How to stop this, especially when the device is not able to send an BUSY back. Use LOCAL channels and parse 'show queues' or

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

Suppose I have one agent login into two different queue and there are calls waiting in both queues. If the calls in one queue has higher call prority (set QUEUE_PRO to higher value) than the calls in other queue, will the agent get the higher prority call first or the QUEUE_PRO has no effect? We have an Asterisk server( 1.2.17 with CentOS) running as ACD. We are having problem using weight option

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

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

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.

Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should "ignore call forward requests from queue members and do nothing when they are requested." Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding

Can anyone explain why when I hit * (as in *2 to transfer) a call that has come to me in a queue asterisk disconnects the call? All I have to do is hit "*" and the call drops.

Hi, On asterisk 1.4.4 I have an strange effect on agents answering queue calls: If an agent does set current call on hold the phone immediately gets connected to the next incoming call. What might cause this effect? How can it be removed? Elmar

Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help.... best regards Thomas

Hi, I have the following requirement.. after a customer is answered by a Queue, I want him to be redirected to another extensions, where an IVR would answer and ask for his opinion about the analyst who just solved his issue. Is there a way to redirect him automatically, or do I have to ask my agents to manually transfer the users to this IVR extension? Thank you Dov -------------- next part

Hi, If an agent doesn't take a call.. is there some way I can PAUSE them instead of logging them out?

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

Has anyone integrated Asterisk Queues with Polycom phones? What I'd like to do is display the agent status next to their appearance. I don't see much discussion about this. This is not the same thing as setting <bw>1</bw> against the appearance in the phone directory. Thanks Doug.

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

Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe

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

Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance