similar to: Asterisk LAMP Developer

Anybody ever managed to implement a solution where one could forward a voicemail from one * server to another? Dominique

I've just seen this post: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is

Hy guys, I cannot understand How the asterisk works. I would like know how the h323.conf, sip.conf and extension.conf works. I don't understand the parameters and the [sections]. What I need to the asterisk get a SIP call and forward them to a H323 terminal. I working at the h323.conf and extension.conf but I cannot understand!!! Please someone can help me. I your can send me a example (with

All, I just had the fortune to take one of the new Senao Wifi SIP phones for a short test drive. First look - it's a nice, compact phone. Weighs around 87g and roughly the size of a Nokia 6210. More on the those later. The thing that struck me was the RF power, it's rated at 100mw (20dBm). That's 10 times more than any of the other brands out on the market Cisco, WiSIP, Zyxel

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Anyone have any experience with either of these, I 'd appreciate some feedback? Plus it seems pretty easy to steal a connection with this. Zyxel Prestige 2000W WiSIP thanks, - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin

Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying

Here's our recent announcement of our new Asterisk Installation CD set: Signate has announced its new Asterisk Installation 2005 CD Set. It's, a complete software PBX (private branch exchange) telephony appliance in a single package. The CD set installs Linux pre-configured for telephony, a stable 1.0x distribution of the open source Asterisk PBX, and Signate's optional, free PBX

> "If" I understood your initial objective correctly (and I may not have), > the user's phones are negotiating the codec to be used for each rtp session. > > Asterisk parameters can be used to dictate rtp sessions between the sip > phone and asterisk, but that won't influence the next step in which the sip > phone negotiates a new rtp session directly with the

Hi all I have just got a P2000w and experience several problems. Hopefully there is someone out there that has got it working. I saw it on Cebit and the person demonstrating it there told me that it was connected to an Asterisk server on the stand -so it should work. Problem 1: it does not register correctly It get lots of messages like this: Jun 26 19:45:19 NOTICE[1107585968]: chan_sip.c:5630

Signate has an immediate opening for a qa/release engineer for our line of VoIP telephony products. Release Engineer Signate is rapidly growing and profitable. We are about to launch a new line of telephone software products. That?s where you can come into the picture. You would support Signate's software development team by reviewing new and changed code, tracking and auditing change

I don't think they are doing it with one Asterisk box. They did say "one rack of servers". Well, that might mean up to 50 computers if they are using blade servers. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of John Todd Sent: Thursday, February 02, 2006 10:21 PM To:

Need new Music on Hold for your PBX? Signate is happy to make a variety of classical music selections available, sampled at rates that are appropriate for telephony. There is no charge. The selections feature Elena Kuschnerova, pianist, and Lev Guelbard, violinist, playing public domain pieces that will give callers a classic impression of you or your company . Click on the link to see a list

I need to dial an extension that tells me what extension I'm dialing from. I'm running a bunch of analog phones off a channel bank to * over a T1. I have the following in extensions.conf. exten => 98,1,SayDigits(${EXTEN}) This says the digits the caller enters on the keypad, not the extension they are calling from. Thanks Guys!!!!!!!! Paul Paul Mahler pmahler@signate.com

i'm getting a tick every second or so on all my calls. All channels are zap channels. Does anyone know how to fix this? Thanks! Paul Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting

Why does voicemail prompt me for an extension instead of just asking my password? [voice-mail] exten => 99,1,VoicemailMain(${EXTEN}@inside) exten => 99,2,Hangup Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?

Hi, I am trying to install a LAMP stack using MariaDB on CentOS 6.4, 64 bit. I have installed MariaDB by using the MariaDB repository configuraton tool. I installed php by itself. Both MariaDB and php individually test fine. I know that php-mysql has to be installed to complete the LAMP stack installation. From lots of googling I understand that there is a php-mysql and a php-mysqlnd.

hi all, i'm not an expert on linux/centos, but i play with it and have a general idea. it's time for me to setup a centos box for development. i rarely install anything from source, except a few times in college when i have to modify kernel for OS project. but i guess i can learn now. i just installed centos 5 with minimal installation. next step is to install LAMP w/ SSL. i

Hi everyone, Do you know any good LAMP developers in Israel who are open to doing projects for people in the US? I have someone who is looking for teams or individuals with experience in e-commerce and shopping solutions as well as content management systems. Mostly PHP, Java, Rails, Javascript, MySQL, etc. You know the drill. If you or someone you know might be interested, please let me know

I see from the archives that Siggi Langauf was wanting to do exactly what I want to do back in November 2003. Here is what he asked: I would like to do a pilot with some legacy gear, however. Accordingly, I'd like to be able to have * dial 1000X where X is the box that has a new voicemail message and 1001X when the user of mb X deletes the new message(s). The dialing should occur