Comments on: Introducing TeleYapper 4.0: The Free, Asterisk 1.4 Message Broadcasting System

By: JMichelin

JMichelin — Mon, 21 Jun 2010 09:27:14 +0000

How difficult would it be to make each group use their own broadcast message? My thoughts were a change to the dial plan logic so that right after authentication you select the group you want to work with then go on the the yapper context, but I am not sure how to maintain ${GROUP} or if it is possible to do something like:
exten => s,4,Record(custom/broadcast${GROUP}:gsm)
or
exten => s,6,Playback(custom/broadcast${GROUP})

where each broadcast file is a concatenation of broadcast . ${GROUP} i.e. custom/broadcast0 for group 0. Does this make sense?

By: John Calderwood

John Calderwood — Wed, 02 Jun 2010 13:51:09 +0000

The greetings are located in "/var/lib/asterisk/sounds/custom/", NOT in "/var/log/asterisk/sounds/custom". Could the posted be edited to fix this? It might save others time. :) [WM: Thanks, John. Sorry for the typo. It's been fixed.]

By: Steve

Steve — Sat, 12 Sep 2009 15:33:56 +0000

Is it possible, that teleyapper leave the msg without ringing the client phone. As there are many other voice blasting service provider offers. [WM: We don't control the end-user's phone. That's under the control of the telephone provider. So... the short answer is "no."]

By: Roll

Roll — Thu, 08 Jan 2009 02:24:29 +0000

Hello, I tried to add the ASNWERTIME variable to have the duration of the call in the 'callee' table, but it appears not to be correctly send through deadAGI. Does anyone have some idea ? [WM: Check for typos. :-) ]

By: Jean

Jean — Mon, 21 Apr 2008 19:16:20 +0000

Great software,

I have one question, what is the maximum simultamnous calls teleyapper can handle?

[WM: That depends upon the horsepower of your server and the number of lines you have available to process outbound calls. Beefy server and unlimited outbound trunks = LOTS of simultaneous calls.]

By: Nick

Nick — Mon, 14 Apr 2008 16:53:13 +0000

yes I’m calling with a pstn line (Digium 4FXO). Hum… Thank you M. Ward for your prompt response but, where I can find this setting (silence duration)?

[WM: nano -w /etc/asterisk/extensions_custom.conf and Ctrl-W to search for Set(TIMEOUT(digit)=4) and Set(TIMEOUT(response)=4). Change the 4’s to 6’s. Then Save your changes: Ctrl-X, Y, then Enter.]

By: david

david — Wed, 09 Apr 2008 17:02:33 +0000

If you are calling with a pstn line and not voip, you need to set the silenc interval to greater than 4 seconds since this is the time between rings (in the u.s.a). Try setting the silence duration to 4250 (4250 ms).

[WM: Great tip! You learn something every day. Thanks.]

By: Nick

Nick — Tue, 01 Apr 2008 20:58:42 +0000

Thank you very much M. Ward. The application work the first time but… The Alison voice start at the end of the first ring, the system doesn’t wait for a answer.

Any Idea
P.S. sorry for my bad english, I’m a french Canadian from Québec

By: varinder parihar

varinder parihar — Sun, 30 Mar 2008 03:12:45 +0000

Where can I get some more information on the text-to-speech feature – I wanted to use this as an appointment notifier and convert the date and time of the appointment to speech. ie Your appointment with us is on and .
Awesome work – love it.

[WM: We plan to release a commercial version of TeleYapper soon that will include the variables you have indicated in addition to other user-defined fields that can be added to your existing database. Stay tuned!]

By: Octothorpe

Octothorpe — Fri, 21 Mar 2008 00:59:29 +0000

Ward – Thank you, thank you, thank you. You have no idea how long I have been waiting for this (OK, maybe you do). Great work as usual.

By: Tom Conklin

Tom Conklin — Tue, 18 Mar 2008 10:55:52 +0000

Ward – I have this app running on my first pass. Excellent instructions!
TomSyr