similar to: asterisk gateway to other gateways

Does anybody developed Predictive Dialer using Asterisk/Digium PBX? Another question: does anybody developed an Dialer using the X100P board? Julio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030718/be051d49/attachment.htm

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the

I have been successfully implementing load balancing gateways for multiple ISP links at various locations using Julian''s patches and as suggested in LARTC HowTo. At one location, one of the ISPs is providing connectivity through a PPOE DSL link which has to be dialled in everytime to connect. The gateway has been configured on a Fedora Core 3 based system and I have recompiled the

Hi to all. I'm trying to use meetme on asterisk 1.4.22.1. On a debian i've compiled (as i need h323 support) openh323_v1_18_0 pwlib_v1_10_0 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-1.4.22.1 All works fine, dahdi status is: asterik:/data/programmi# /etc/init.d/dahdi status ### Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: RTC) 1" (MASTER) asterik:/data/programmi#

Dear all, I am really new in this world, In my office i setup a Asterisk and all extensions are working fine . i also add some trunk to route my local calls. As some user needs to call overseas so we are looking for cheap gateway with SIP supported. Now . we found one gateway . they just give me there softwitch's ip and told to me to send my call there. i have no idea how to router my

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

I have a friend traveling overseas. I want to allow him to call a number which will give him a busy signal (so no charge), but will then send me an email that he has called. I know how to use a call file to trigger a call (I created a callback system for myself when I traveled overseas a few months ago), but I don't know how to trigger an email. All I want is a simple message like

Hi everybody. We have a samba4 domain deployed across serveral countries. Some of them (overseas) have a poor VPN connection with mainland. Since Samba4 does not support (yet) subtrees, we have deployed a DC in each location for domain validation. However users in mainland logon randomly at overseas location and sometimes this is a problem due to low bandwidth available. Is there any way to

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

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

So I'm not getting caller ID via the X100P card. I've confirmed the PSTN is sending me caller id by plugging in a little third party box. Ideas, tips, greatly appreciated

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

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

Hello all, I was wondering if anybody knows where one might obtain a PCI ISDN card supporting a single BRI for use with Asterisk in Australia (and using something like chan_capi). Because of the Isdn4Linux DTMF issue, I don't want one of those cards. I've already spent too much time messing about with my current card. I'm after something like the AVM Fritz! cards. I found one place

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

I'm thinking of doing an app to work with the CID that's gotten from the Zap channel. All the CID's I see from within the US are 10 digit numbers. I'm out in the rural areas of the US, and no-one ever calls me from overseas. If they did, what would the CID look like? What does the CallerID look like overseas? How many countries provide it? murf

First, please forgive me if this is a total newbie question, I've only just begun to scratch the surface of asterisk. I currently have a dialplan set up to let me dial a specific extension, authenticate the user, then have * dial a hard coded/programmed overseas number. What I would like to do is set up my dialplan to have an extension that offers up an outbound dialtone allowing the

Hello, I would to allow users to place calls overseas such as India and Malaysia but only with a security code. if they don't have a security code I want to be able to drop the calls. can someone point me to a right direction to achieve this goal? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone. I have the following problem that could help would appreciate. 1) Environment: HOST 1: xen-hypervisor-3.2-1-amd64 with all VM in LVM disk. HOST 2: XCP 0.5 2) The VM that I wish migrate is a domU in XEN and is debian lenny over LVM: ................................................................................................... # Configuration file for the Xen instance