similar to: SIP REFER: Trying again

Hello, I have the following scenario: 1. VoIP Gateway G400 connected to PSTN 2. Asterisk server 1 (working as IVR) 3. Asterisk server 2 (working as ACD, with several agents connected) I have incoming calls coming from PSTN through the VoIP Gateway to Asterisk server 1 (IVR). When the IVR ends working with the call, transfers it to the Asterisk server 2 (ACD). In Asterisk server 1

Is there an easy way to automatically log agents in? We are using the queuing function to front end a main number without really using multiple agents. The downside is during a restart, or system reboot someone must remember to log in the agent. If I could incorporate it into a startup script it would be much more convenient. I've done some looking around and see references to

Hi all- I have a customer with a Rockwell Spectrum ACD. They wish to connect to an asterisk system using euro-ISDN circuits, and asked me if asterisk supports a feature that they call "Take back and Transfer". This feature allows an IVR (asterisk in this case) to handle a call and then blind-transfer it back through the ACD to another extension- freeing up the asterisk channel.

Hello, does anybody know about an existing skill-based routing solution for asterisk? I found only some theoretical documents on voip-info.org. I would like to have finer control over who can get which call in which order. Example: Several operators with several topics. Each operator may have a given knowledge-base for given topic. Topics may be weighted in question of complexity as well. Some

Hello, We use and develop the astGUIclient suite. It is Open-source(as in GPL) and offers Inbound and Outbound call center functions with reports, ACD, monitoring, recording and very basic IVR scripts. Complex IVR functions need to be custom programmed within Asterisk but that is not really that hard. It works across multiple Asterisk servers and we are using it currently at 5 locations including

Ok I have tried to post to this list server but have just gotten the automated reply saying the moderator has to approve it to the list first which was my mistake for sending from the wrong email account. So if the moderator finally approves my questions and you see the same post again "Sorry". My situation is this: I havn't installed Asterisk yet but am curious the general way

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 all, first post, need help. I'm running a complex asterisk 1.8 install with five machines. I inherited it and don't fully understand it, nor the deep mysteries of asterisk either. I would appreciate any insight you might have. I scoured the 'net and the Digium wiki and my Google-Fu has failed me. I've been asked to somehow prohibit transfers to extension 3232. It has to be

HI, theres a lot of controversy related to this topic, my company is thinking on me to take the astricon bootcamp, but want to know if it is really whorty, 3000 USD is a huge amount of money to spend, plus the hotel, food and transportation, ive already deployed some asterisk?s pbx and have experience with it using analog tdm cards and E1/T1, queues, conference rooms, IVR, ACD, inbound and

On Mon, Jun 9, 2008 at 7:29 AM, Hendrik Boom <hendrik at topoi.pooq.com> wrote: > I don't know if the toy program in chapter 4 of the tutorial > implementing Kaleidoscope in llvm with C++ is part of your > regression suite It isn't (although that might be a good idea). > but with the version of llvm I installed > last weekend, it does not compile: > > hendrik

Hi Hendrik, > hendrik at lovesong:~/dv/llvm/tut$ g++ -g toy.cpp `llvm-config --cppflags --ldflags --libs core jit native` -O3 -o toy > toy.cpp: In member function ‘virtual llvm::Value* NumberExprAST::Codegen()’: > toy.cpp:359: error: no matching function for call to ‘llvm::ConstantFP::get(const llvm::Type*&, llvm::APFloat)’ > /usr/local/llvm/include/llvm/Constants.h:237: note:

Hello, I've seen various suggestions thrown around for hardware when people ask, but can we all agree on some basic hardware recommendations for a few basic setups(and post them on a website) to make it easier for new people to avoid some of the hardware/software pitfalls when they are setting up their first systems. Something like this: (THIS IS JUST A PROPOSED LAYOUT SO PLEASE BE GENTLE)

? Hi, I am working on a small inbound call center solution that uses an ACD system. I might add an IVR system later on. I only have 2 extensions set up (extensions 1000 and 1001), I want the system to put new calls in a queue if both extensions are busy. I am currently subscribed with a SIP trunk provider and can successfully recieve calls. I want?to design a system where customers?can call my

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

I don't know if the toy program in chapter 4 of the tutorial implementing Kaleidoscope in llvm with C++ is part of your regression suite, but with the version of llvm I installed last weekend, it does not compile: hendrik at lovesong:~/dv/llvm/tut$ g++ -g toy.cpp `llvm-config --cppflags --ldflags --libs core jit native` -O3 -o toy toy.cpp: In member function ‘virtual llvm::Value*

On Mon, 09 Jun 2008 21:25:42 +0200, Matthijs Kooijman wrote: > Hi Hendrik, > > It seems the ConstantFP method get might have been modified to take a > reference-to-APFloat, meaning you can't pass in APFloat(Value) anonymously. > You should either put it in a local variable, like > APFloat F(Val); > return ConstantFP::get(Type::DoubleTy, F); > which I _think_ is the

IPManager is now fully customizable as everything is generated from templates using the IPManager database. Complete set of templates is included in the download. You can configure the dial plan and other configuration files exactly like you want, using nothing but notepad. There's a very easy syntax for retrieving data from the IPManager database. You can even have different templates for

Hi Group, Is there any information available for Asterisk 1.8 dimensioning? I googled but couldn't find helpful data for 1.8. I am trying to figure out hardware configuration for following features implemented in Asterisk 1.8? (1)100 SIP clients. (2)ACD (Around 15 realtime queues) (3)Call recording for all SIP clients. (4)4 port PRI (E1). There would be around 100 concurrent calls.

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. > > >

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