similar to: Re: Asterisk-Users Digest, Vol 6, Issue 284

Hi all, I have a question regarding ACD for queues. What happens when I have 2 or more queues with same weight and each queue has a call in? How will it decide which call will be routed to the next available agent? Will it take the call with the longest waiting time in queue? If not how would I do this? Also can someone point me to resources for making a single queue with customer calls

Hello! I am new to both the list and to "*". Can someone direct me to some documentation concerning the reporting tools available for use with "*" as a call-center system? Specifically, things like ACD offer/taken, wrap-time, and such? Thanks very much. This looks like an exciting project. I'm looking forward to playing with it! -- Michel R Vaillancourt Avaya

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

Hi all, I have a need where ACD ringall and ACD roundrobin ring strategies will be combined. Basically, ring every agent in a specified order, but whenever it times out and goes to the next agent, I still need the previous agent(s) to continue to ring. I would like to develop this extension myself as a contribution to the asterisk community. In doing this, where should I start? I've

Hello - I have just joined the lists and am considering installing quite a few * systems. I am looking for an IP-PBX with both solid standard features and call-center/ACD features. I have read the documentation and the list archives and did not see any references to real call-center type reporting and queuing. Is anyone out there using * in this kind of environment? The features I would be

I'm having trouble getting the quota plugin to work with deliver. The quota in the database is not getting set. I am using dovecot-1.0-rc15. This server is not yet in production. Some values below are because I'm trying to test with the simplest possible case, and some because it's reading from a different database. Also, I noticed that "dovecot -n" doesn't display any

Has anyone had any luck getting the Avaya IP SoftPhone to work in wine? I get these errors on Ubuntu 8 Code: preloader: Warning: failed to reserve range 00000000-60000000 wine: creating configuration directory '/home/microchp/.wine'... preloader: Warning: failed to reserve range 00000000-60000000 preloader: Warning: failed to reserve range 00000000-60000000 err:dosmem:setup_dos_mem

See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >

Now that most of you have worked overtime to show why most people are continually pissed at Nix Users (all except two of course). The problem I can see is the downright technosnobbery involved. There is nothing wrong with Linux. I play around with RH9 and FreeBSD and find that most things run fine. But you get into a problem where it keeps asking for the same blamed libraries over and over on

Hello, I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have done this before between Asterisk 1.6 and Avaya but had some issues placing external calls from the Asterisk to the Public network which is connected to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is outdated and has no support. On the Asterisk side I have Aastra 6731i SIP phones

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --

----- Original Message ----- > Subject: Re: [Asterisk-Users] Analog Phones with Status Light Indicators > From: Adam Goryachev <mailinglists@websitemanagers.com.au> > To: asterisk-users@lists.digium.com > Organization: Website Managers > Date: Thu, 12 Aug 2004 14:53:02 +1000 > Reply-To: asterisk-users@lists.digium.com > > On Wed, 2004-08-11 at 20:42, Steven

Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf

I'm trying to connect an Asterisk system to an Avaya Partner ACS R6 system. The problem I'm having is that I cannot get the partner system to get CallerID over the T1 modlue. The partner is using the T1 with E & M signalling (which I don't think can be changed), and whatever I tried didn't work. My only option right now is to get FXS ports on the Avaya side plugged into the

Good day, Our installation needs a robust ACD application (as I'm sure others do) that can be dynamically reconfigured (if possible) maybe by a MySQL database. I have looked at Bill Heckel's ACD work and Andreas Otto's DynExtendb as well as James Sharp's ACD. None of these seem to be quite finished. Is there a chance a unified (* blessed) ACD application can be put

Hello Sir, I have few clarifications as we are planning to work with asterisk. If you don't mind, please clarify the following:- Q1. Do Asterisk support ACD functionality? If Yes, can you give information on how to configure or work with ACD (and it's usage). Q2. From the list of features listed in www.asterisk.org , I see "Predictive dialler" is listed

Hello, We have a mixed environment here consisting of a number of Avaya PBX systems, a group of Cisco Call Managers, an H.323 gateway on a Cisco router, and an Asterisk server. The PBX land is connected to the VoIP land using the Cisco router/H.323 gateway. The Asterisk system is running code from the CVS tree from around mid Oct 2005. The OH323 driver on the system is from inaccessnetworks,

Hello! We have asterisk connected over PRI no our phone network, so I'm avaya PBX user. Asterisk connects to another avaya system over h323. Connection can be shown as avaya--PRI-asterisk--h323-avaya When I do call as avaya user I see name of remote end avay user, i.e. connected line info. As I see in debug remote side send is as 14:07:29:758 Received H.2250 Message = { 14:07:29:758

Christopher- Nothing like defining a complicated environment. I do have some experience in this arena- but unfortunately, not with the OH323 driver- I generally stick to the Nufone driver, as I find it more reliable overall. YMMV. One thing that might help is if you could tell us if it ever worked, or if this is a new problem that's cropped up since a particular change. Still- there are

Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) "[Nov 6 17:14:23]