similar to: FW: [asterisk-dev] Hardware and CentOS tweaks.

Hello All, I have a requirement to setup a predictive dialer for a customers call center. I am asking for pros and cons of the different dialers available for Asterisk. If you are going to send marketing material send it to my e-mail directly please and not to the list. I was hoping to get the opinions of any one using any of these dialers and what they liked and didn't like, ease of

I have a customer with a more or less unique need. Right now we are using Wombat as a dialer software so they can contact clients for QA purposes. Everything is working very well and their contact center productivity is way up from the old manual dialing method. The only thing we are having a problem with is that they have up to 5 phone numbers to contact a single customer. Obviously

Currently I have a predictive dialer that is web-enabled, as well as a superdialer mechanism. I was wondering what kind of success people have had with their dialers (lessons learned, etc.) and how I need some direction on how to seamlessly integrate an inbound system with the predictive dialer. My thoughts are that inbound phone calls will be routed to a receptionist first, who will then

Does anyone have any experience using asterisk with an external predicitve dialer, like MediaTel? Specifically: The predictive dialer dials out over T1 circuits. It connects to asterisk via amphenol cable from an fxs card in the dialer to asterisk with a tdm2406 fxo card. In the analog world, the dialer dials out through the t1 circuit, and the fxs card is plugged into a 66 block so the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U)

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

I am looking to see if anyone has any info on auto dialer software that connects directly to a voip provider without using any third party boards or digium cards? I've been building dialers for the past 5 years and I want to get out of using add on cards and just make calls from the software directly using voip. The software would need reporting features, answering machine detection, hangup

Hello Guys, Last Sunday I had the idea to expand my gameplay btrfs. Now what was missing was a yum plugin which worked like arch or opensuse. Ok, I told myself and learned a little python. Also something for grub2 to boot into a snapshot. My results are up on: https://copr.fedorainfracloud.org/coprs/andybe/Centos7Btrfs/ grub2-btrfs (do not use grub-btrfs, this is the version before)

Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM "Customer Relations Management" with an SIP dialers built in.the question is the following.. is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will

Hi Tilghman!   > Clearly, you missed the point.  Since there is a FREE predictive dialer out > there, and your product costs something, you are not the world's cheapest > predictive dialer.  I respect your wording and the way you or other people think on the list about difference between cheapest and free predictive dialer.   Surely

HW = HPQ DC7700 DuoCore 1.8GHz, 4 x 2GB RAM, 1 x 1TB SATA (AHCI) OS = CentOS-5.2 ix_64 (xen) I am experimenting with Xen and with virtual machines in general for the first time. In consequence I have many questions. However, the one that I wish answered at the moment is simple: Given a DVD containing the CentOS-5.2_Final (i386) distribution mounted at /media/CentOS_5.2_Final on Dom0, what

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

I have a problem that has developed within about the past 3 months with my backup outgoing SIP provider (I am not sure when this problem started since it involves only my backup provider which is used rarely). The problem is that most (not all) outgoing calls fail during the earliest stages of call setup, specifically after the provider sends back a "407 Proxy Auth Required" response.

​I need to launch a remote process at the machine that has the dialer. I could hard-code the IP address in a global variable, but It would be much more elegant if the dialplan would have a "manager" object where I could read "manager-->connected". ​ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am using the most recent stable icecast (2.4.3) with the following config: https://gist.github.com/mspanc/5e560e8fbd07bf65bc30a288cb514c14 From time to time, CPU usage hits 100% and keeps on that level once it’s hit. According to htop, only 20-25% is a regular usage (green bar on htop), rest are kernel threads (red bar on htop).   Am I doing some relaying and fallbacks as you can see

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

One of my user is using asterisk 1.4 based Dialer i.e Vici 2.0. Today, when they downloaded , the CDR from the carrier site for 26th June 2010 , they see 50% calls are NEVER dialed by Dialer but it appears in CDR. Amazingly, all the call durations are of 29-30 secs. When we checked the status of the same in Dialer, lead is present there but its marked as NEW which means Dialer has ever dialed

Hi, I was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian.

Hi I just upgraded my system to xen-3.0-testing, hoping it would solve my problems with the virtual network interfaceses (I cannot connect between two domains) and memory usage (I sometime "miss" a few hundret megabytes). The update didn''t help on the networking issues, but it did on memory usage. However, I created a new problem: using htop, the cpu is used at 50% in both