similar to: Asterisk-Users: Termination

Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the

Piling on... InterNIC says the domain was created almost a week ago, and expires in a year. The registrar is GoDaddy. The owner of the site is located in the Dominican Republic: C/1ra #15 Costa Criolla, Km9 Carr. Sanchez Santo Domingo, New York 00000 Dominican Republic Registered through: GoDaddy.com, Inc. (http://www.godaddy.com) Domain Name: CDSPORTAL.NET Created on: 14-Mar-08 Expires on:

Hi, I've been reading all I can on Google (and Asterisk TFOT book) looking for ideas on how to implement an automated blacklist feature. I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be

Hello: I would like to know if there's a way to request free chanels from remote asterisk servers ? My idea is to make an agi returning a dial to inter-asterisk connected servers when there's not enought chanels on local server, maybe like a ping to all of them or maybe requesting to a central server where all the *s send and request information about available chanels each 2 or 3

Hello-- I've been playing with the privacy options on my home/home-office system since August last year, and have some results, gleaned from my CDR records, which over the last 13 months, number a total of 8672, which includes incoming, as well as outgoing calls. Before I start spitting out numbers, let me note that with the current setup, I haven't had to tell a single telemarketer

Following new features are now supported by asterisk based telemarketing software Auto subscription / registration after call recipient press a key in voice broadcasting https://www.ictbroadcast.com/Subscription-Campaign-to-automatically-register-customers-at-website-with-Voice-broadcasting-Autodialer There will be restriction to call a number in off time accordingly to timezone of

Hi, Returning to my office I find two "missed calls" (from autodialers) that my IP601 displays as originating from 01111111111. However the CDR database recorded the call this way: calldate: 2008-04-04 14:18:16+02 clid: 0172752780 src: 0172752780 dst: 2131 dcontext: default channel: Zap/1-1 dstchannel: SIP/0146472131-007a7e80 lastapp:

Hi all, <p>once again i came up with my conferencing stuff. On a conference with more then two people it's a waste of bandwidth, that every entity send it's data to every entity. Since there is only one audio line, the audio must be merged on the server. Here are my questions: - How many audio chanels can a server process (let's say a 3GHz machine) in this way: decode all

Hello, I've got two streams, one for broadband, one for dialup. Well, having had occation to use a dialup connection recently i checked the dialup stream. Although it was streaming what the broadband stream was, the audio quality was audibly worse. It didn't buffer, but it didn't sound as clear as the broadband stream. I used lame to encode the tracks to mp3 and used it's

Does anybody know why I get this error when using zapateller: WARNING[1209214400]: File rtp.c, Line 327 (ast_rtcp_read): RTP Read error: Resource temporarily unavailable It's scrolls until a sound is recived from the line, then it plays the zapateller tones. /Mike

Hi Everybody I've a requirement for one of my operators for an autodialler for which i plan to deploy asterisk (I already have 3 asterisk servers on PRI running very well ! ). The scene is like : Asterisk will call a customer and play a prompt that prompts him to press 1 if he wishes to talk to an agent , If the customer presses 1 then the call gets connected to one of my proffessional agents

Hi, please excuse my poor englisch. Is it possible to connect a (privat Test-Asterisk) to my privat ISDN and allow him to only answer one dialed number? We have 3 up to 10 Numbers on each (Euro-)ISDN (2 b-chanels), it cant't be done by the last Digits cause the numbers are completely different. For Example: I have 3 Numbers (641717, 928752....) Is it possible to tell Asterisk (in

Hello List, I'm working on an autodialer project. At the moment I use the Originate application then I "throw" it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate finish its bit and I have lots of wasted time or even worse, the remote party hanging the call because instead of a human

https://bugzilla.samba.org/show_bug.cgi?id=8856 Summary: --hard-links does not handle hard-linked symlinks correctly on FreeBSD Product: rsync Version: 3.0.7 Platform: All OS/Version: FreeBSD Status: NEW Severity: normal Priority: P5 Component: core AssignedTo: wayned at

I did some digging and found a thread from 2011 talking about how to subclass Xapian::PostingSource in order to incorporate the date or recency of a document in its weighting: http://thread.gmane.org/gmane.comp.search.xapian.general/8849/focus=8856 As in that thread, I want to be clear that I don't want to sort by date, but rather incorporate date information into the score by which I

I have been bugging by a telemarketer who does not take any cue at all. So I look up the Asterisk Handbook and send his call with the respect caller id to my voicemail. Has any one implemented any of this feature with database for more caller ids to be included?? David Kwok

http://bugzilla.mindrot.org/show_bug.cgi?id=938 Summary: "AllowGroups" option and secondary user's groups limit Product: Portable OpenSSH Version: 3.9p1 Platform: ix86 OS/Version: Linux Status: NEW Severity: major Priority: P2 Component: PAM support AssignedTo: openssh-bugs at

I am pretty much a newbie at CentOS, and Linux on client side, I would like to help out occasionally by helping the web development crew in their endeavors and by sorting out some stuff. such as repairing broken links, or just scanning for content and maybe even doing a little bit of consulting. Eric R. Clark Dynamic Technology Systems US Phone: (817) 704-4109 US Cell: (817) 706-8856 Email is

I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling are all our customers, etc. so I don't need to do any do-not-call checking. Just

I have a * serving 15 sip clients. I use the digium 4 port t1 card. We have an autodialer that calls and reminds clients of there appointment. it uses a pri t1. I would like to plug its t1 output into asterisk to use voip. I am very new to * and am confused. Any help would be appreciated. _________________________________________________________________ Express yourself instantly with