search for: askterisks

Hi, I'm trying to plot bar graphs with stacked columns marked with askterisks for certain columns, which follow certain criteria. I've gotten the stacked bar plots with Data set A (please refer to code below). However, I haven't figured how to put asterisks on columns based on some extra information (such as "used/unused") as in Data set B i.e. ast...

I have: RedHat 9.0 TDM40B asterisk-0.9.0 compiled from sources zaptel-0.9.1 likewise /etc/zaptel.conf contains fxoks=1-4 loadzone = us defaultzone=us loaded modules zaptel and wcfxs /etc/askterisk/zapata.conf contains [channels] language = en signalling = fxo_ks context = phones channel => 1-4 /etc/askterisk/extensions.conf contains [general]

Hey guys, Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Ive read the Success stories form voip-info.org but

Hey all, Does anyone know what this means? I was running asterisk fine. Installed it on a new pc and I am using the g729b. codec that is optional. I ran the install for the codec it went ok but when I run askterisk via asterisk -vvvgc it gives me this error anyone know? I make sure I entered in the correct reg number. I followed the steps correctly. Too Registration error! Please try

Hi everyone, I just installed asterisk on my system with the purpose of rerouting calls on sip channels. I don't think i need any hardware for that. I am using LIPZ4(zultys) and sjphone as softphones. I tried setting up both of them and to call one from the other on the same machine, however could not. I 1-) I could connect sjphone in isolation to freeworld dialup howver i got no sounds

When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call

> > I guess my big question is: is it possible to have extensions mapped to > > people, not to phones? > > Yes, you just need to link the user/extension to a technology/channel > when logged in, and to a bogus value when not so that your dial will > fail quickly and fall through to voicemail. Also you will want to make > sure your voicemail goes out in email since you

hi all, this message is destinate to users who rub libunicall on asterisk succefuly ... my first question is: what is the version of linux you use to make asterisk run with the new libunicall? and what is the version of asterisk? my problem is as follows: with version FC3 I have success to run libunicall without problem, the testcall run perfectly well. askterisk run without problem on the

Hi, I've been trying to get my zaptel x100p cards working for the past week now. this is what I've done: installed asterisk: make clean make linux 26 (for fedora core 2) make install installed zaptel: make clean make make install did a modprobe zaptel, and wcfxo got this in /var/log/messages: PCI: found IRQ 11 for device 0000:00:0f.0 wcfxo: daa mode is 'FCC' found a

Dear Steve and *.* e1r2 developers and users; my first question is: what is the version of linux you use to make asterisk run with the new libunicall? and what is the version of asterisk? my problem is : with version Fedora, C3 I have compile and run libunicall without problem, the testcall.c run also perfectly. and askterisk run without problem on the localm network. I patch asterisk in

Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u