search for: astericks

Earlier this afternoon I had this working exten => 2890,1,Answer exten => 2890,2,GoTo(12) exten => 2890,12,Wait(1) exten => 2890,13,Festival('I can say numbers like') exten => 2890,14,SayNumber(1230001,f) exten => 2890,15,Wait(1) exten => 2890,16,HangUp I was so very proud of myself... All of a sudden after a reboot.... I get the following from the same call plan

I am looking to help out my company find a more budget conscious but reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have high speed internet. Currently we dial up an AT&T conf using regular analog phones. I don't have a great grasp as to what Asterick is capable of, but my

I am running Asterisk CVS-HEAD-07/14/04-16:28:29 and noticed that when I send a subscribe I get back a 403. This used to work in an old version which I forgot to record before upgrading to the above version. Any suggestion? I can register fine with the * server. Sip read: SUBSCRIBE sip:2486@192.168.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK46F2668 From:

I am having trouble locating webvmail on my * server. Is this a seprate porgram or does it come with *. I am running version asterick*CLI> show version Asterisk CVS-03/26/04-17:08:20 built by root@localhost.localdomain on a i686 running Linux asterick*CLI> Thanks Kurt __________________________________ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25¢

Hi All... Can Asterick detect distinctive ringing on a POTS line and answer with different configurations? Thanks...

Patrick hi. Asterisk can do that, and you don't need VOIP lines. If you connect Asterisk to the net, and all employees have a VOIP phone (either hardware or software) then you're good to go. What do you need? To begin with, install linux on an old pc (well, not too old). Then go to voip-info.org and take a look at the Asterisk wiki. Everything you need is there. And of course, we're

I've got a fresh FC2 install and I'm trying to get the symlinks right according to the /usr/src/zaptel/README.Linux26 instructions. I've created two symlinks: /usr/src/linux-2.6 -> /usr/src/linux-2.6.5-1.358 /lib/modules/linux-2.6 -> /lib/modules/2.6.7-1.494.2.2 When I do a "make linux26", I get a million warnings and errors with the result being: make[2]: ***

...o 192.168.1.2 etc. 2) VOIP to someone outside in the U.S. 3) VOIP to someone overseas e.g. U.K. 4) Get a hardware card for the incoming line 5) Some extensions, perhap's Four (4). I don't see much in relation to point 1. What software (linux) can be used to connect VOIP to the astericks server ?? Regards...Martin

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

Folks, My question concerns the SIP Notify that is being sent to ... device. You can see it in the following line: Voicemail: 0/0 Shows no Voice mail but I did leave a voice mail at the extension. Any suggestion on what I should look for in my * setup. I am not worried about the 481 coming back for the other side yet. Once I get a handle on the Notify, I'll work on the 481.

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show peers *CLI> sip show peers Name/username Host Mask Port Status 3014...

I'm reading the Asterisk book, 2nd edition, and it describes how one can set up voicemail to be delivered by IMAP to a voicemail folder. Asterisk can monitor flags on the folder so that the "message available" light on one's phone tracks the state of the "read" flag in the folder. One can either dial in for one's voicemail or listen to it from one's favorite

...rameters as a single character string. Now I know that sys.calls()[[1]] will give me the full text of the initial call, but the problem is to detach the ... above from that as a text string. If I could do that I'd be done. Here's the incomplete code with comments -- see the gap set off by astericks. rStd=function(x,...){ if(missing(x)) # must have a specified function { cat("Error: No function specified\n"); return(invisible(NULL)); } z=as.character(substitute(x)); # must include code here to check that z is the name # of one of our alte...

...t even write protected. The only readme file was 'README.DevKitLite' and sure enough, it was an explanation how to install the dev lite kit, which I DON't have. I tried it anyway ignoring the references to S100U. There is a astinstaller, that failed complaining of no openssl-devel. Astericks has been running for the past week with no obvious problems or alerts. I am informed that mandrake has renamed it to libopenssl-blah-blah, so I look at the astinstaller entry and change it. Still fails so I remove that check. It now builds and completes. It complains [root@carol etc]# ztcfg -...

First off, I have to admit that I'm uncertain if this is the appropriate forum; I'd be happy for suggestions about where else to look. I'm doing this work on a stock install of CentOS-7-x86_64-Minimal-1810.iso, with no updates. I'm trying to create an RPM database from a custom set of RPMs. One RPM ('openldap-ltb' from the LDAP Tool Box project (ltb-project.org) has a

On Tue, 12 Feb 2019, Brian Reichert wrote: > First off, I have to admit that I'm uncertain if this is the > appropriate forum; I'd be happy for suggestions about where else > to look. > > I'm doing this work on a stock install of CentOS-7-x86_64-Minimal-1810.iso, > with no updates. > > I'm trying to create an RPM database from a custom set of RPMs. >

I get this error no matter what I do. I've tried switching to different locations on the TDM400p to no avail. I've got the card setup with the green module in slot1 and red in slot2. My config files are exactly like the "Configure Dev-Kit PCI (TDM11B)" example. http://www.digium.com/index.php?menu=configuration#TDM11B When I run ztcfg -v, I get: 2 channels configured.

Hi? 1. When a mobile user dial an IVR short code , mobile network able to divert that call to Asterisk platform.? 2. There would be web servers which are holds Voice XML . 3. Asterisk would be able to redirect the mobile request to certain Voice XML server accordingly. Just for like this setup , how do we install asterisk with minimum of asterisk modules ? Do we need to install complete

Hi If I do not use any DAHDI - hardware , can I ignore the DAHDI linux and tools installation ? Can I use just asterisk ? is there any dependencies while executing Asterisk with DAHDI modules ? Thank you Luke -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130607/b7c8e748/attachment.htm>