similar to: #include not working in 1.4

I'm trying to make an AMI call. I want to call a number, play an announcement when the call is answered, then call a second number and connect the two when the second call is answered. I an able to make a simple call to two numbers and connect them using the manager API but playing the announcement has me beat. Suggestions anyone? Bruce Ferrell

Hallo everybody, today i've released the very first (and very alpha :-) version of ABCTI, a CTI solution for Asterisk written in python, using pyGTK and py-Asterisk. It talks directly to the Manager API, so no additional software is required on the Asterisk server. You can find it on Sourceforge as http://abcti.sourceforge.net Basic functionality is already there: Originate, Transfer,

hi, when i type ******** asterisk-1.4# ./configure ****************** i got this error **************** configure: error: C++ preprocessor "/lib/cpp" fails sanity check See `config.log' for more details. ***************** # vi config.log *************************** ...........; cpp: installation problem, cannot exec 'cc1plus' : No such file or directory ..............

Here is the complete output of ztcfg: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected It appears that none of the zaptel devices have been created. I did not notice any errors during the make install. Does anyone have any suggestions? I know the hardware works, because it was working as my asterisk 1.2 test

Does anyone know where I can get an rpm spec file for zaptel 1.4.x? Thanks, Doug. ____________________________________________________________________________________ Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. http://mobile.yahoo.com/sports;_ylt=At9_qDKvtAbMuh1G1SQtBI7ntAcJ -------------- next part -------------- An HTML attachment was

Don't you just hate it when something was working and when you come to use it in anger it's broken :-( Something in the, fairly, recent series of Asterisk updates has broken DIGITAL call passthrough. I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover cable). This PBX used to be able to

Hi all, I thought I'd share I script I made (based on some of Leif's works) that lets you download, compile and install Asterisk all in one go; and then removed the dev tools used. We use it quite a bit to provision systems using Ansible, but it is easier than remembering everything every time even if you are using a shell. At the moment I have scripts for Centos 7 and Asterisk 13, but

Is it possible to include other context under globals? what i wanted to do is create a new file which have other variables on it because i don't want to edit every now and then the extensions.conf just for the global variable. -- Regards, Mark Quitoriano http://asterisk.org.ph -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am using asterisk1.2 In this, my dialplan is going large , so i need to configure this small pieces for this, i did in my extensions.conf when I dial the 123 its not going , means that file is not reading. is there any parameters to add any where ? please tell me this #include is not working ... extensions.conf [general] [global] trunk=zap/g0 #include exten-internal.conf [default] exten

Hello again list, I'm running a 1.4.42 install on SUSE with an AEX410P card. The DAHDI release is 2.4.0 because the machine won't properly install 2.5 and also won't install Asterisk 10.0 because I can't get a good SQLite3 library to install. Whenever I enter DTMF very quickly or very slowly, app_read des on me. Has anyone experienced similar joy

Hello everybody! I have some problems with my Astersk. I have an analogical OpenVox card and A Billion ISDN card (with mISDN). I load the modules with modprobe zaptel and modprobe wctdm. When I run ztcfg -vv I have this: Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No

Hi All; If I need to set the opermode to King Saudi Arabia, what the name I have to use? For example, to set it for kuwait then I use opermode=KUWAIT. So what will be for Saudi Arabia? Any advise? Regards Bilal

Hello Friends, I am trying to Installl dahdi on amazon EC2 which have Open-SUSE-11.1 X86 version. and here is snap of uname- a command *Linux ip-10-160-86-41 2.6.32.19-0.3-ec2 #1 SMP 2010-09-17 20:28:21 +0200 x86_64 x86_64 x86_64 GNU/Linux* when I try to run DAHDI distribution dahdi-linux-2.1.0.4 I am getting following error *echo "You do not appear to have the sources for the

Asterisk 1.4.19, Zaptel 1.4.10 and libpri 1.4.3. When I try to start up Asterisk (with -cvvv) I get an error regarding func_curl.so (lines omitted) ... == Registered custom function STRFTIME == Registered custom function STRPTIME == Registered custom function EVAL == Registered custom function KEYPADHASH == Registered custom function SPRINTF func_strings.so => (String handling

Hi, Does anyone know how to include a file in AEL using the #include "filename" syntax in .conf files? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B

Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683&nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev

I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I can see the channel bank with lsusb, but when I tried to use zaptel_hardware, or when I try the /etc/init.d/script, they don't see my Channel Bank. I compiled the latest Zaptel 1.4.10, with Astribank's dependecies, fxload and libusb-dev. Anyone have a similiar experience ? Best Regards, -- Guilherme Loch G?es

I have one TDM2404E digium card on asterisk box, after configuring the zaptel and zapata configuration files, I am getting these errors when reloading asterisk: ast_unregister_indication_country: Removed default indication country 'us' setup_zap: Ignoring signalling setup_zap: Ignoring answeronpolarityswitch unable to recognize channel 13-5 what is the reason for that? Thanks,

Hello Has someone written a web page (preferably PHP) that simply shows what extensions are currently online? Thank you.

With one IVR payment system, I noticed quite a difference in DTMF transmission between these two cards. The IVR missed nearly all digits from X100P, while receiving digits from TDM fine. Since neither card process or synthesize audio, what can the difference be? (This particular IVR has problem with some regular phone devices, too.) Yuan Liu