similar to: Newbie voiceplus + asterisk

Anyone using asterisk with VoicePlus ? Looking for a new provider and wanted to get a little feedback

Hey guys, Cal someone help me. I'm register voiceplus DID i try to config fllow example but not work. When i test call to number and debug iax2 in my asterisk not found packet. My iax.conf -------- register => in-xxx:yyy@gw5.voicepulse.com [voicepulse] context = voicepulse-incoming secret=yyy auth=md5 type=friend host=gw5.voicepulse.com ------ extention.conf ---- [voicepulse-incoming]

We are looking for another solution besides Samba, which will allow our application developers using Windows 95/98/NT to access share on our Sun Box running Solaris 2.6 operating systems. Our supervisor feels the Samba installation process is more complex than what we really need. Any suggestions? Steven Hawks NCR Corporation IT Services - GAD Sales and Business Infrastructure * 1529 Brown St.,

Ladies and Gentleman, I have setup Asterisk and AMP. They are working with out error. But now I need to get everything going. I have Voicepluse and they give sample iax.conf and extensions.conf files but that does me a little good as I am using AMP. Is there some docs somewhere on how to confgure once I have AMP up and running? I am not a telephony guy and alot of this looks like greek to

Hi, I am trying to get the extension status (weather it has dialed outgoing call via SIP or IAX2), using the following piece of code however it always returns -1 on all the extensions (valid/invalid). Am i missing something ? Any help. Thanks ----------------------------------- #!/usr/bin/perl use Asterisk::Manager; use lib './lib', '../lib'; $|++; my $astman = new

Hi, I'm getting an issue while executing AMI Originate. I'm getting "extension does not exists" on Originate's Response, and on the other hand Asterisk CLI say "fwrite() returned error: Broken pipe" Please suggest me what is wrong. Muhammad Faheem ### my originate code block ...

hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [CC]

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Hello , I have a problem to install asterisk with ldap. I am doing the following: make clean . / configure make menuselect LIBS =- lldap export LIBS make ====> This is where my error #make CC = "cc" CXX = "g + +" LD = "" AR = "" RANLIB = "" CFLAGS = "" make-C menuselect CONFIGURE_SILENT = "- silent" menuselect make [1]:

Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use 555@mainmenu and 666@mainmenu) for the Icon extensions. IAX softphone seems to work

Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a

Hello All, My build fails in the "make asterisk" section, here is the log of the final steps it performed. All help appreciated... Thanks and Regards Ajit ============================================ make[1]: Leaving directory `/usr/src/asterisk/cdr' make[1]: Entering directory `/usr/src/asterisk/astman' rm -f *.o astman make[1]: Leaving directory

hello i am getting this error while trying to run ast-rad-acc.pl my $ast_connected = 1; while( 1 ) { if( $astman->connect ) { $ast_connected = 1; syslog('info', 'Connected to Asterisk!'); $astman->setcallback('DEFAULT', \&status_callback); eval { $astman->eventloop; }; print STDERR "Connected To Asterisk\n"; } else {

Hello experts, to avoid any unknown problems with my Linux installation I have now as a last resort method installed SuSE Linux 9.0 a new and have downloaded a fresh copy of Asterisk via CVS. Then I followed the steps of the "Getting started with Asterisk" and compiled successfully zaptel and libpri (as far as I can see). But when I compile asterisk I get an error. I have attached the

Hello, I am trying to setup an Asterisk GUI with the help of astman(please visit http://astman.sourceforge.net/am-user-guide.html). I have installed astman and currently assessing my GUI using; http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl I am trying to get the menu options in my GUI to work but to no avail. Currently my parameters are set to; Asterisk Install Directory:

I got the error below while compiling asterisk. Please offer me some help. hubert for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x depend || exit 1 ; done make[1]: Entering directory `/etc/asterisk/asterisk-0.7.2/res' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/etc/asterisk/asterisk-0.7.2/res' make[1]: Entering

Hi everyone, I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel 2.6.9-42.0.10.ELsmp) and am having a lot of trouble getting asterisk running on it. I had a fair bit of success with the ATrpms binaries (Zaptel worked but asterisk failed to startup because it couldn't find the speex modules). I am trying to thus recompile the asterisk rpm for CentOS 4.4 with the least amounts of

Hello ppl, Have implemented a really nice Billing engine using AGI scripts. So far it works fine, tho haven't yet put it in the torture cell. The AGI scripts have been written in PHP, using MySQL for the billing and profile information. The major disadvantages I see using AGI scripts : 1. A new process(invocation of PHP scripts) on every new call. 2. MySQL connections on every instance of

has any body considered using astman/gastman to show a) sip/iax etc registry status with other * b) manage multiple * boxes would this be any use, i was just thinking if some tech support has to handle multiple * for one organization, it may be worthwhile to have mutiple * boxen become part of one REALM and manage that realm, or maybe some other way, but i'm sure we can find someone to

Newt is a separate library from asterisk that is used to easily create a ncurses based program. If you want to see some examples of how to use newt, look under the 'zaptel' source for zttool.c and/or under the asterisk/astman source for astman.c Newt pretty easy to code off of and can make a quick & easy frontend for things. Check out the /usr/share/doc/<newt directory> for more