similar to: Anyone using VoiceMaster

Hi all I have been using * for a couple of month and now I wanted to install the Asterisk Calling Card. But when I try to install I get the following error: [root@terraserver1 astcc]# make install mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi >/dev/null Detected dry run! ./astcc-admin.cgi

_____ From: Habiyakare Aimable [mailto:aimable@terracom.rw] Sent: Monday, June 07, 2004 11:49 AM To: 'asterisk-users@list.digium.com'; 'gt'; 'support@digium.com' Subject: Problem with Asterisk PRI forwarding to SER Hi all, I have a problem. We have a phone system setup like this: SIP phones------------>SER--------------->Asterisk---------------->PSTN(PRI

HI all, Any of you having experience with voice master? I tried using the openh323 channel it doesn't give me voice at all. THere's no packet coming in. There's no problem with any other equipment but voicemaster doesn't send voice at all. Funny thing, i have an old version of OpenPhone, it's working. So please if any of you knows this problem, please share. THx a bunch

I'm working on calls coming in to an asterisk box as H.323, and going out as SIP to a remote device (a VoiceMaster). The remote device is refusing the calls with SIP error 406 (Not Acceptable). I have attached the SIP debug output below. It looks like codecs overlaps - can anyone see why the call is being refused? (Note that I'm not registering with the remote SIP device, just

Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on

Hi all, I need to connect two asterisk server in same network and i'm using sip user as my clients...... plz anyone suggest me.... Regards, Thiru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061227/aa4e409c/attachment.htm

Hi all, Is there a way of disabling ALERTING message on a PRI channel? I have a problem .* is sending ALERTING message to the Nortel telco switch of my local provider BEFORE it dial the number it has to .if the number is busy or invalid there is no way we can tell this to the switch because it has already been told that the phone is ringing I am using asterisk Asterisk CVS-04/06/04-10:46:21 with

Hi all, I need help. I have a Linux box with SER as a proxy server with ip phones attached on it , and another linux box with Asterisk and T410 card connect to an E1 line .Whenever there is a call from PSTN it is passed to Asterisk and then to SER box and then to the phone .every time an invalid number dialed from PSTN to SIP phones connected to SER asterisk says that the call is progressing

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

My SIP-provider sends my a SIP-invite like this : INVITE sip:329298yyy6 at 80.XX.XX.69:5060 SIP/2.0 Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c Max-Forwards: 70 From: <sip:321445xxx6 at 80.XX.XX.69>;tag=f395877e02bf8eb2fd8f5a0e To: <sip:329298yyy6 at 80.XX.XX.69> Call-ID: f395877e02187250fd8f5a0f at 80.XX.XX.68 CSeq: 1 INVITE User-Agent: SysMaster VoIP

Dear all, I have installed asterisk 1.0.5 on redhat 9 I have installed also, asterisk-oh323 0.6.5 module (successfully compiled and installed) Now When I am trying to get asterisk communicate with a Radius (in my case: it's the VoiceMaster Radius) I was able to do the following: After installing all recommended to download and install radius client for asterisk

Hello, I want to implement a perl logon script which would map network shares depending on group membership. This way when I move a user to or from a group it automatically gets the new shares. The PDC is Samba 2.2.1 or 2.2.2 on BSD and clients are NT workstation and 2000 Pro. Perl is latest activestate for win32 and is intended to run on Win32 clients. The big trouble is getting user Group

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

Folks! I dont know whether anyone has done this exercise before of putting together a Wish-List of things that you want to do, if you have all the gadgets you need and have a client base that needs Asterisk's Features and more. Here are some of the scenarios I am playing out that I will do, once I have enough time. Can anyone add to this list of Scenarios with or without using other gadgets,

Hello Guys, After adding money into my sysmaster phone account I am able to make calls outside.thnx _____ From: Mani Nair [mailto:mnair at nvloisp.com] Sent: Friday, September 07, 2007 9:16 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Sysmaster and Asterisk Hello Guys, I am unable to make calls to outside number from some of my extensions.

Sherwood, Your intentions are noble and your desire to build this, fullfills an immediate need for business. If your intention is just to build a GUI for Asterisk, read no further. If your desire is to build something more purposeful, your best bet would be to see the existing commercial GUI/HostedPBX offerings like Pbxware and Switchware from bicomsystems.com ( http://www.bicomsystems.com)

Hi, Have a problem in my extension.conf: I have: [sip] exten => _333.,1,wait,3 exten => _333.,2,Answer exten => _333.,3,AbsoluteTimeout,7 exten => _333.,4,Hangup I wanted to test if * is executing this dial plan by calling 3335254255 for example. The problem is as follow: It waits, it answers but it does not seems to see the Absolutetimeout: call goes forever. What's wrong? Am

As far as I can tell from <http://shorewall.net/errata.htm> the current shorewall bogons file is <http://shorewall.net/pub/shorewall/errata/2.0.8/bogons> which contains the line: 58.0.0.0/7 logdrop # Reserved This is incorrect. These two /8s were allocated to APNIC as of April 2004. See also <http://marc.theaimsgroup.com/?l=nanog&m=108319003517919&w=2> and the main

Gday all, No adapter materializes under Windows XP Service Pack 2 ! Latest OpenVPN drives install, but tinc can not handle this and does not work. One would assume that tinc has this problem solved by now ? Any updates yet ? Regards, Jan -------------- next part -------------- An HTML attachment was scrubbed... URL:

? is there any speex code in place that may help enable variable speed playback ? or perhaps another open source project has some code for this ? variable speed playback for voice messaging is an attractive feature, that would be well recieved for sure ! thx for any reply, Mark Tom Harper <tharper@sightspeed.com> wrote: At 12:35 PM 10/6/2004, Matthias Granberry wrote: >There is