similar to: INFO: TFOT book- n priorities and labels

I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. -- Dave Cotton <dcotton@linuxautrement.com>

If i get a 8XX number, my provider told me that they will send all the calls he gets. But due to bandwidth and asterisk capacitiy in this particular installation, the system is only able to handle 27 incoming calls. How in my dialplan do I regulate, sending a busy signal, when my system hits 27 incoming calls? thanks, -- ------------------------------------------- Erick Perez

Or you could just use "System" and do it the easy way! AGI is better suited for more complicated scripting. In any case, check out the TFOT book - you'll like it. -MC _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Collins Sent: Thursday, February 09, 2006 3:13 PM To: Asterisk Users Mailing List -

Hello, Thought people might be interested in this. I want my voicemails emailed to a person and not stored on my asterisk server. However, I want them to have a sequential number. I found that if I set the option delete=1 in my voicemail.conf file for the mailbox, then the numbering would keep being restarted. I wrote this shell script to fool Asterisk into numbering my voicemails sequentially

Seri, I think you might just need to find the right variable and pass it to your script. Also, you'll probably need to read up on AGI - I'm assuming that's how you would launch your Perl scripts. Here's the list of variables on the wiki: http://www.voip-info.org/wiki-Asterisk+variables Most likely you're looking for the variable referred to as ${CALLERID(num)}

Hi! Im looking for a very basic example for the following simple problem. I've been searching voip-info.org and looked in the ORA book without a clue. I have a SIP account at sip.provider.com and my own asterisk server. What I want is the following: I. Register my phone to my asterisk server, not directly to provider.com II. My asterisk server should ring my phone when somebody calls me

Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel

Hi: When should we use "unnumbered priorities"(n) in extensions.What is the different between these 2 forms of extensions.conf? and ,Are both true? extensions.conf: form1: [Conferencerooms] exten => 333,1,Answer exten => 333,n,meetme(8000|cim) exten => 333,n,playback(vm-goodbye) exten => 333,n,hangup form2: [Conferencerooms] exten => 333,1,Answer exten =>

I know this one must be easy but I'm an newbye so please help. In my extensions.conf I want to have a line like: Exten => 9XXXXXX,1,Dial(Zap/4/${SOMETHING},40,r) Ie: I want to dial all the XXX-es, but not the 9; How do I do that? What do I write in place of ${SOMETHING}? Navigating the wiki didn't provide any usefull advice... Thanks.

I am new to asterisk. I need help getting started, if it's even worth getting started. I say if it's worth getting started because I'm not sure if my hardware will even work with asterisk. I have a US ROBOTICS 56K V.90 PCI SOFT MODEM. I have standard twisted pair telephone wire. I can't afford to alter my hardware. I know I won't be able to do any sophistocated VoIP stuff.

Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming

Is there a list somewhere or a way to find the following: 1- All non US 48 area codes which can be dialed as 1+10 2- All strange area codes which are used for premium services such as 900-XXX-XXXX 3- Anything else that should be restricted if one was to restrict all calls to US 48 only I have found many list but it's tough looking at the entire list of area codes and pulling out each of them

ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! ClueCon was put together by Asterlink, the same team of people who helped shape Asterisk into what it is today by writing features, fixing bugs, offering IRC support and assisting with the management of the development effort. We have produced several real-world solutions based

One thing I forgot to mention: I also cropped the registration and cut marks off the sides of the pages. If you want the uncropped version, get: http://www.alexburke.ca/asterisk-tfot-uncropped.pdf Sorry about the excessive noise, but I figured I should mention this. >Date: Mon, 20 Feb 2006 18:55:50 -0500 >To: asterisk-users@lists.digium.com >From: Alexander Burke

Any ideas? Sangoma support is closed for the evening. I have the latest Sangoma drivers and Asterisk 1.4 everything installed. When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up" repeated over and over. The B channels never come up. There are no errors in any of the logs, zttool, or the wanpipe tools. Intense pri debug output: < Unnumbered frame: < SAPI:

Hi I have a small ISDN PABX , (BRI) at home - Siemens Gigaset 4175 which has cordless DECT extensions. I have set up asterisk on FC3 with two HFC cards and I am using the latest bristuff. I am trying to use * between my ISDN line and my PABX. One card is in TE mode and can receive and make calls OK. I can make and receive external calls to a SIP phone. I cannot get the other card to talk to my

I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 numbers routed through it. When the calls come in, I get the following message on the console and the call never makes it through: (800 number is fake) Extension '8005551212' in context 'nonauthenticated' from '232102749585' does not exist. Rejecting the call on span 4, channel 1. I do have the

Robin, Thanks for the viddler.com suggestion! I'm uploading all of the ClueCon videos to it right now. John, so far I'd have to give viddler.com two thumbs up. I'm adding my stuff here: http://www.viddler.com/explore/cluecon Your ClueCon presentation should show up some time on Friday. I've noticed that there's a little bit of a lag time between upload and video being

I'm relatively new to Asterisk and I'm hoping attending Cluecon will be a good way to get up to speed on the project and hear about what others are doing with it. We currently use a Cisco IP phone system at my office, although I just added an asterisk box to provide soft phones to our travelling users. (IAX2 is a lot easier to get through firewalls than cisco's protocols). Terry

List, Does anyone know how to add the dst Country to the CDR's via Macro (preferably). For example, I will add a column in the cdr DB table and when someone dials 01158212XXX. I want the CDR's to show Caracas as the destination in this new column. I have all of the International destinations in my extensions.conf like the example below: [macro-dialout-intl] exten =>