similar to: VM access

I subscribed to this list for about two months before I began posting, so I've got a buttload of email to sift through ... I'm doing this BEFORE I flood the list with my inane questions ... But here goes: I read a reply from one guy to another about recording. The message included this context from extensions.conf: [recordings] exten => 500,1,Festival('Please record your

*I have a Ham Radio program, named CQ100, it works fine using WINDOZE, but when I installed the same program on my linux system everything works except there is no-audio I'am using Ubuntu 7.10 linux... The author told me that windoze uses GSM 6.10 codec, plus ACM audio compression manager, these are built-in... So by anychance does anyone know of a program that one can get to use on a

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

Hi, I have been trying to workout how to use the call pickup. So Far, I have the following in zapata.conf [channels] signalling => fxo_ks context => local pickupgroup=1 callgroup=1 channel => 1-3 When I dial *8# all I hear is busy tone. What have I missed? thanks Jay.

>-----Original Message----- >From: Larry Shields [mailto:LJ.Shields@Verizon.net] >Sent: Friday, August 27, 2004 12:20 PM >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe() >If I assign the DID to ring extension SIP/2000 and then after time-out send >it to MeetMe() or Playback() it works and the caller

Hello, I'm a newbie to the asterisk system. I'm trying to configure a dialplan so that when I use my IAXy it will prompt me with an IVR and then send me off to different things like dial and voicemail from that. I've tried various combinations but I can't seem to get it to work properly. Here is an example: [default] exten => s,1,Answer exten => s,2,Ringing It gives me

I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

Hi, in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? Also I want to know if there is a option that erase all message in a user box. Best REgards Ever Zalazar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060512/98a6f962/attachment.htm

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

Folks, I've just successfully set up Asterisk (as part of the Asterisk Management Portal installation). When I say "successfully", I mean that I have gone through all the steps detailed for the installation of AMP and not hit any snags there. I can connect to my asterisk server via ssh and can also connect via Http to the portal to change settings in AMP. Now I'm trying to

I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)

*I've been looking into installing the latest version of Compiz-fusion 0.7.6, also the latest plugins... Plus any how to install info... Does anyone know a site to automatically update it, from what I now have that came with 8.04 version of Ubuntu... Thanks for any help would be appreciated... Larry * -- Powered by Debian/GNU/Linux by Ubuntu ver 8.04 Hardy Heron 73 de Larry/wd9esu

When a person calls into the Asterisk voicemail or auto attendant, the first second or two are cut-off. This happens with custom prompts I have created (with or without 1 or 2 second delays) and with the default prompts that come with Asterisk. Does anyone have a solution to this problem? I'm running the current CVS. My default menu config is: [mainmenu] ; ; We start with what to do when a

I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? TDM800P and HPEC The TDM800P is an 8-port analog telephony interface card, so it fills the gap between Digium's 4-port and 24-port cards. Analog phones and POTS lines are going to be with us for some time, and demand for support for them remains high. The

For anyone playing around with IAXy(S100i) devices, I am making the following available: Windows IAXy Provision v1.00 This is a from-the-ground-up development of a means of provisioning IAXy devices using a Windows environment. For some users, being bound to Linux for IAXy provisioning is not viable or convenient in some cases. This application provides a GUI data entry for the various IAXy

Hi.. I have to setup an extension in a remote location that will use a cordless analog telephone.. I am looking at the IAXY to do this for me..Basically the data path will be as follows... [Asterisk] == (NAT) == {Internet} == (NAT) == ATA -- Handset Since there are two NAT boxes in the path I know SIP won't work.. I also don't want to move the Asterisk box to the internet side of the

Hi, I have three Iaxy devices (s101i) parts. Two of them seem to work fine. The third plays a loud repeating click sound when an analog phone is plugged in. I can provision all of them, and make calls to all of them. The clicking one will blink when a call is incoming, but no audio from the call can be heard on the handset, and the caller only hears silence. The same handset works on the

We have many IAXy devices in the field now. In all cases, in iax.conf, we have "qualify=yes", so that using "iax2 show peers", we can see whether or not the device is currently online. In some cases, the IAXy device and/or Asterisk are not communicating their qualification, because "iax2 show peers" shows the device as status UNKNOWN. However, when a user picks up