similar to: pri to voip

Hello, I'm looking at building a couple new PRI Gateway boxes using TE405P cards, and was wondering if anyone has had any experiences (good or bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics builds some really nice (and cost effective) 1U servers based on the board: Server: http://www.gtweb.net/gt637.html Specs:

> We once developed a reminder system for a customer. He's a cleaning > company, cleaning homes and offices. He was spending two hours a day calling > his customers to remind them of their appointment the next day. Two hours a > day equates to 40 hours a month that he saved with that system. He's been > using it for maybe 6-7 years now and not once was a customer upset

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

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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 > On 2/02/2017, at 9:52 pm, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > > <snip> > > but in simple solidarity with everyone who has ever > > been pissed off by a machine-initiated spam marketing phone call at an > > inappropriate moment, I am not going to tell you how to do it. > >

I'm in the process of writing a simple autodialer to dial a list of numbers and play a message. One of the options I want to give them is a way to "dial X to have a customer service representative call you" Looking for a simple way to pass the number that I dialed to a script in extensions.conf... something like this: [serviceinterruption] exten =>

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

Greetings all, My next step is to purchase a nice VoIP phone for my desk. I have a grandstream, and the sound is great, but I'm looking for more of an office style phone, preferably that can handle multiple lines, has a more flexible display (i.e. name as well as number). SIP would be preferable. Any suggestions? Thanks, Eric

Hello, I want originate a call to some destination, and when B side answes to play a prompt. Asterisk version is 1.6.0.5. But also I need to insert a SIP header to Invite, that's why I'm using Local Channel. This is my extension.ael: context autodialer-local { _X. => { SipAddHeader(P-Asserted-Identity: <sip:${CALLERID(num)}@xxx.xxx.xxx.xxx;user=phone>);

Hello, I'm currently developing a simple asterisk application using SFS (Skype For SIP) which tries to call to an outbound number, play a message and read DMTF digits. My first approach used the Manager to originate calls and then called an agi script to deal with the rest. Anyway, this ended up being not so clear because the call did not start on the Originate extension that it was supposed

Not sure if anybody noticed, but it seems like Skype For Asterisk is out. $66 per channels, pretty pricey http://store.digium.com/productview.php?product_code=1SFA0001 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090817/cd8c6546/attachment.htm

<http://www.vistaprint.com/vp/gateway.aspx?S=5176697856> Anyone on the list good with Linksys PAP2NA configuration, I am looking to take my ata's and emulate the operation of a pots phone line as close as I can get. One thing I need to change is the fast busy tone I get when someone hangs up on the call. We are also looking for a Voip/ Asterisk Consultant to set up hardware for a

Hi, Is it possible to use 4 TE405 boards in one server ? It mean, to have 16 E1s on just one server. Can somebody tell me how many boards is it possible to have on one server ? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060605/29e9d42d/attachment.htm

Hi all, I'm planning to develop a custom autodialer application which will be dealing with its own model for agents and queues, therefore it won't use neither asterisk agents nor asterisk queues, nor asterisk cdr. The application will supply the whole reporting and agent managing features by itself. The application will command asterisk through an AMI telnet connection using only the

Hello ! I 've some questions how bridging of ISDN calls is done. Assume an asterisk system with a TE405 card equipped. (PRI1 - PRI4) An incoming ISDN call on PRI1 is transfered back to PRI3. Unless there is DTMF detection or other things involved, the bridging is done without Asterisk. Does this card have a some sort of cross connection ? Does the PCM leave the card ? Or is there some DMA

I have a TE405/TE410P card that was working on 1.0.X I upgraded the OS to Centos 4.5, Updated asterisk to 1.4 and zaptel to 1.4.5 and libpri. I copied all the zaptel and zapata and extensions.conf files from 1.0 I did update extensions.conf from 1.0 to 1.4 commands. I cannot get the card to work in 1.4.10. AHHH! I see with zttool that the T1 is in Green, I see calls coming in as the bits go

http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view Enjoy!

Hello, I have a client that has a fairly small installation (20 SIP Phones) that is running Stable. Asterisk appears to be consuming large quantities of memory, and growing uncontrollably to the point where after about 6 weeks the box starts to swap itself to death. I've been keeping my eye on it today, and in the last 12 hours, it has grown by about 8 megabytes, and there has been

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial