similar to: Pass Dialed number to a script

I'm trying to get a script working on a website to send out automatic email reminders to customers reminding them monthly to change furnace filters. I haven't got one running successfully, yet. That made me think - could it be done with a phone call using Asterisk? A monthly automated phone call to remind people to change their furnace filter? I have no ability to figure this out

Hi All, I have been struggling with the auto dial out in asterisk. I am trying to get a call to be auto dialed and play back a message once the line is answered. So far I have been unsuccessful. Currently what happens is I have my .call file. I mv it into /var/spool/asterisk/outgoing. The call is initiated and that all works, my problem is that it does not wait for the line to be

I'm trying get this working. I've looked through the list, and can't see how to get AMD to print out more. I have it call and say Hello like I normally would. I've tried to say more and less doesn't seem to matter. After I hangup it does recognize hangup. Here's logging during an attempt where I make outbound call and answer, but then hangup after 1-2 seconds: Jan 24

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

There was a post in the 'wiki' for an application to provide an outcall when there is a voicemail is left on asterisk. I am having a problem that this application will only work if the caller presses the pound sign at the end of recording. As most people just hang up, this application isn't working. Can any offer suggestions to accomplish this out call?

<FONT face="Default Sans Serif, Verdana, Arial, Helvetica, sans-serif" size=2><DIV>I have been playing with getting the sample.call file to work by dropping it into /var/spool/asterisk/outgoing.&nbsp; The process works to the point of calling the desired number and plays the message.&nbsp; The problem is that the message starts playing almost immediately, so if the

Hi Bastian, I'd like to propose backporting the following changesets from upstream xen-unstable into the xen-3 package. With these it is possible to boot a default installation of Squeeze (using d-i) in a domU using pygrub. 20480:c2c2e67b8198 pygrub: if default entry is "saved" then use first entry. 20481:8f4e0adc2b3b pygrub: expands tabs before displaying menus. 20485:086a6a0c3f37

Hi All when I try to use auto-dial to connect to outside phone , my applications get executed before the caller attend the calls , this happens only when I call outside no , ie when I use Channel: ZAP/1/0507451111 in my sample.call file , if I use Channel:SIP/326 , it works fine my ?sample.call? file contains Channel: ZAP/1/0507451111 Callerid: Asterisk MaxRetries: 2 RetryTime: 10

Hi, I am trying to insert a system() call right after the call to Voicemail() in order to notify people that they have received new voicemail. (In case you are interested, I was following the tips here, but I have had to change a few things: http://www.voip-info.org/wiki-Asterisk+tips+callback ). My setup works when someone leaving a message hits # to exit the voicemail after speaking, or

hallo, i trying to dial with a python script via the manager interface, it works ok but i would like to send a soud file name as a variable to the dialplan, so that i can call a number and send it a different soundfile i choose in my pyton script. the problem is, that the * dials correct and sends a sound but only if its hardcodet, the variable my script sends will not bee seen in the

IPSwitchBoard Version 0.69 has just been released; it is available for FREE: <http://mambo.thorben.dk> Download here Release notes: * Record calls by right clicking any extension button, you can have several recordings at the same time. * Bug fixes The recordings will be placed as a single wav file on the Asterisk server in the folder: /var/spool/asterisk/monitor the name of the

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

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

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 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

Hi all- I have asterisk running on 2 systems, with four E1 spans each. Each system is connected to a (big) NT DMS-100 switch. For load testing an IVR system running on one of the asterisk systems, I'd like to use the other system to generate a lot of outbound calls under program control - on most or all of its channels simultaneously. All of the asterisk dialplan and agi programming

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

Hi, We are in the process of moving from an Avaya Definity to Asterisk for our institution's phone system. I got one feature that the Avaya had, which I have not been able to reproduce with Polycom phones and Asterisk; since this feature seemed so small and useless to me when testing, I kind of ignored it. Now I am getting more "I miss that" requests than I expected. =) On the

I am trying to retrieve the "dialed peer number" but it seems that ${DIALEDPEERNUMBER} is "broken". Also, I know that I could extract the dialed number from the ${CHANNEL} variable but this only works for SIP and maybe IAX (untested). However, it doesn't work for ZAP. All I get when using ZAP is something like "Zap/1-1" (for SIP I would get