similar to: PHP Web interface for Asterisk

Hi everybody, I've been tinkering with a web based interface for Asterisk. I tried to stick as closely to the current configuration format as possible. The web interface should help to do things a little easier (sort by extension, context, do bulk changes). www.packetbell.com/asterisk Feedback appreciated! Dylan.

I would like to integrate * with an existing Altigen PBX. I want to spend as little money as possible to make it happen. My main goal is to inexpensively connect a branch office to the phone system. Eventually I would like to replace the Altigen system with an Asterisk server so I don't want to spend any money on Altigen hardware. Currently the Altigen has analog interfaces with a couple

Has anyone got Asterisk talking successfully to an Altigen PBX using h323? I can successfully make calls from Asterisk to Altigen, but calls from Altigen to Asterisk get a fast busy. There seems to be a lack of h323 example files (or maybe I'm looking in the wrong places) as well as a severe lack of h323 documentation from Altigen. Any pointers would be greatly appreciated.

Hi Everyone. I have a client that uses an Altigen system. I am really new to PBX systems so all this is totally foreign to me. They currently have 5 inbound trunk lines and about 20 analog phones. >From what I can gather they are using the Altigen Quantum cards that support 8 extensions and 4 trunks. >From what I can gather the solution is a TDM04B and TDM01B to bring in the lines from

I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that

I've tried timing faxes two ways:

OK let’s start out with this. I’m not a pro GUI designer… ? Now that that’s done. Welcome to OpenConf. At least that what we call it now. To config an * file click on the filename to the left. For my example use extension.conf. Now you’ll have a FULL text editor and a parsed list of all the [sections] in the extensions.conf file on your left. On the right you will find any numbered var’s

has anyone done caller announce in MeetMe's before? Dave P >>> brian@bkw.org 5/18/2004 5:50:49 PM >>> With multiple parking lots you can give each person their own lot thus exten 800 for everyone will connect them with just their call passing the lot name which you know for X customer. bkw ----- Original Message ----- From: "Andrew Kohlsmith"

I have put my phpconfig stuff out into the Digium CVS tree. Project name is phpconfig. see it at http://rads.netcom.utah.edu/phpconfig/phpconfig.php Lemme know if you have any patches or add on's are welcome Dave Packham aka p0lar

I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server. When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the

does anyone have a recorder GSM file that emulates the Telco's "if you are a telemarketer please hangup now" recording? I don't see one in the sounds dir. the ZapATEller works great for computerized callers but if a human hears this message asking them to go away they have to. Isn't that right? Dave

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but

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

Hi, Can anybody explain me in configuring Asterisk to record a conference? Regards... Girish _________________________________________________________________ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now!

we are in process of writing a PHP interface for * conf files. we are parsing the files like INI files but the only prob I have so far is that separate extensions in a context dont have any unique tag that I can capture. This works ok [trunkld] ; ; Long distance context accessed through trunk ; exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}) exten => _91NXXNXXXXXX,2,Congestion

Hi everyone, I'd like to use Asterisk to build a phonetree (www.phonetree.com) type of application, like this: 1. Read a text-based name/phonenumber file. 2. Call every number and play a recorded message. 3. If a beep is detected, replay the message from scratch (to leave messages on an answering machine). 4. Write results to a log file. Does anything like this exist already? Can this be

I've been digging around with some cisco engineers for about a week & I finally got an encouraging response to the Auto-Answer issue with the SIP Phones. Here is their reply: =============== == FROM CISCO == =============== Auto-Answer feature is introduced in SIP IP Phone 6.0 version. This software version is expected to be available for customers shortly. Please let me know if you

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

Before you get too far.... The internet line jacks dont allow outbound calling. they cannot be used as trunk lines to the PSTN. the outbound code has not been written yet. I had to go buy FXO card from digium (that works much better than the Linejack) to get outbound calling to work Dave >>> fplandae@hotmail.com 4/11/2003 5:35:24 PM >>> Hi, I am a newbie. I have been

Hello all, I seem to be having a problem importing a data set from Excel into R. I'm using the "read.table" command to import the data with the following line of code: > newborn<-read.table("newborn edit.csv", header=T, sep=",") where "newborn edit.csv" is the name of the file. Unfortunately, I'm getting back the following error message: