similar to: Linksys PAP2NA Configuration / Asterisk / Voip consultant wanted

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, Anyone here have experience with Audiocodes MediaPack MP-108 Gateways? I would be willing to pay someone for advice and support with configuring my gateways for a telemarketing project I am starting. My experience is somewhat limited but all I want to do is make outbound calls just like I would on normal pots lines. (That's the best way to explain it) I do not need any special

Hi People, I'm writing to get some advice on where to start when learning asterisk? I was going to begin learning with AAH but I wanted to find out if there is a certain build to avoid or if there is a Gui/front end that is better then another. I have been working with dialogic cards for the past 5 years and with auto dialers but I want to get into providing voip service, support and

Hello all, I am trying to find out if anyone has a provider that is good with dtmf playback using a Sipura 2100? I've just dialed with voxee and the call goes through but when I press 1 my dialer does not " hear" it. My dialer is making the call using a Dialogic d/4PCI connected to the Sipura 2100 through voxee and I am calling my landline. When I pick up the landline

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

I wanted to get everyone's opinion on an issue I am having. I am currently using linksys PAP2NA ATA adapters to terminate analog calls from my auto dialer to the voip termination co. The problem I have is when I call the PSTN everything goes fine until the person being called hangs up the phone. Once they hang up on the PSTN side it takes almost 15-20 Seconds for the ata to see the

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>);

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

Does anyone know how the Linksys PAP2T ATA's can be mass provisioned? Documentation seems to be sketchy, even on the Linksys web site. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070607/3f90695c/attachment.htm

Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the

Hi list! I have an Asterisk server (1.2.14) connected to a E1 line via a TE410P, and some PAP2NA connected to it. The PAP2 DTMF configurations is set to INFO and Asterisk to INFO too. At first, is INFO method different from RFC2833?? Well, I have two problems. The first is that when I place a call to outside, via E1 trunk, sometimes I get some DTMF tones and I'm sure nobody hit any key. Seems

All, I am looking at a little support on this, as I haven't found it on google yet. I have had this work on Callweaver, but am moving to Asterisk for a variety of reasons. My dial plans, and everything else transferred perfectly, though I am not sure they are 'correct' for Asterisk 1.6.1, with simple things like SIP users outlined in the sip.conf file, not in the users file,

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

We''re a J2EE development shop in Vancouver, BC, Canada, and are taking a serious look at Ruby on Rails for some major projects. We''re looking for a consultant with extensive (5+ years) J2EE application development experience, who has deliverd one or more Ruby on Rails projects. Initially we are looking for someone to provide 3-5 days of on-site consulting to work with our

Hello- I have a client in Orange County California who will soon need some consulting assistance with their new asterisk system. I've been asked to help them find someone. Skills needed would be, in order of importance: Basic experience configuring and using asterisk, coding experience in Perl, experience with MySQL or equiv., and a knowledge of telephony terminology and

We are trying to install a Tellabs echo canceler on our T-1/PRI and having no luck. We can't get the lines out of AIS/RED/BLUE alarm when we put in the echocan. We are looking for a consultant (near New Orleans) to assist us in installing this device. Contact John DeMajo <jdemajo@stirlingprop.com> at +1-504-858-7689 --Eric -- Always do right. This will gratify some people and

Hello All, How are you all doing today? Good I hope. I am sure that I have asked this question before, but recently lost my emails server and thus any replies that you may have sent me. We are working to get a small online VoIP service established and I am looking for someone who might like to partner on this project or possibly offer reasonable consulting services. We need someone to take the

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

I'm trying to define distinctive rings for lines in this gateway but don't works. Nothing happen when sending a call...the phone doesn't ring.... The same configuration works fine for PAP2NA devices. Adriano Almeida Flickr agora em portugu?s. Voc? clica, todo mundo v?. Saiba mais. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 =>