[Asterisk-Users] IVR only system with scalibility with asterisk???
Hello all: Thank you for taking the time to read this post. Background: I am a new user to IVR systems and asterisk. I have been tasked with helping to set up a system that will only handle IVR (eg no PBX functions) incomming calls for 45 or so people that will call in 3 or 4 time each day during (approx) normal business hours. We have started to look at the Ivrs perl module from http://search.cpan.org/author/MUKUND/. We are having limited success. I found the asterisk software and have trugded through the last several months looking for IVR specific comments with minimal success. Issues: 1. We need to have a working system by "yesterday" (since we were told yesterday ;) my problem not yours). Realy, how easy is asterisk to develop for in a IVR message -> response -> authorize/validate -> contiune scenario? We will need to do database lookups. 2. We expect that we will end up greater than 100 users that will call in 3 or 4 time each day during (approx) normal business hours in the next couple of months. We also have the possibility that the next step may involve several hundred users. How can I provide something now and scale UP from a "commidity" PC (running GNU/Linux of course)? The Wildcard X100P only has 1 port. Are there other higher density options that "just work"? I've seen mentioned an Intel/Dialogic card that looks high density and expensive and interesting. I don't mind having a farm of these things on commidity hardware... within reason. Again, I'm a newbe trying to get myself up to speed on this topic. 3. I need to provide a working model very soon. What is cheapest way to put together a system with AVAILIBLE parts? There seems to be a shortage of some of the cards on the yahoo store front. So, looking for everything... Thank you very much. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR only system with scalibility with asterisk???
Top posting only: This is great info. A couple of you have already replied with very helpfull and usefull information. Thank you very much!! I am very excited to hear that I can test without purchasing the hardware. I googled and found a IAXClient at http://iaxclient.sourceforge.net/. Is that the program you mean? It looks like * is a very good sofware to pursue and very powerfull (and fairly inexpensive) when hooked up with the Digium cards. I will download and begin trying *. I will likely just place an order for a single analog card just to get the ball rolling very soon. I still would like to hear more about how people are integrating * with external scripts. It was mentioned that the docs may be a little sparse... examples would be GREAT (said in the voice of Tony Tiger). On Thu, Sep 04, 2003 at 01:15:11PM -0500, Steven Critchfield wrote: > ...my original post deleted > First you need to decide on how many ports you will need, how important > ease of scalability is. For the number of ports, you need to decide how > much tolerance you have for the people remotely to deal with a busy > signal. So far you mentioned 45 people making 3-4 calls a day over a ~8 > hour day. The quick math says that 45 people with 4 calls is 180 calls a > day. In a 8 hour day you have 480 minutes. From 480 minutes 1 port could > handle the load if the call was under 2.5 minutes long and everyone > waited till it became available. My guess is you don't want people on > redial that often and waiting for the port to come open. Next, you move > on to what is the acceptable idle amount of service available. If you > scaled up to say 5 lines, and the call length is short, then you will > have your service mostly idle, but it can handle peak times better. I'll > let you continue this line of questioning internally. > > Next to decide on hardware, if you think you may need more than 10 > lines, you need to move to digital trunks. You can start with a T100P > and a channel bank until your costs justify switching over to a T1. The > benefit is already having the hardware in hand and used to it while on > spending a little more short term to get the FXO channel bank that you > will either sell off later, or convert to FXS for internal extensions if > you want to switch services. If you already have a PBX in house and can > drop a T1 interface to you asterisk box, that is good too. > > As for your application. You mention looking into perl modules, so I > assume you have some perl familiarity. From AGI you can script up any > database access and prompting you so wish to undertake. Essentially it > will come out to be something like. > > stream file(prompt) > while (not enough digits) > wait for digits > collect dialed digits > validate(digits) # in this sub is where your database stuff works > continue? # whatever here you planned on letting happen. > > > all this is easy and cheap. For your quick demonstration, I suggest > setting up asterisk with a dummy interface, downloading the iaxclient > and showing that your AGI app would be easy enough to write. You are > then only into the project for time, but not any parts. Once you have > that down, you would then purchase the parts needed to complete the > project from Digium and deploy. > > If you stick with a T100P interface then you should be able to handle > 500 people with 5 minute calls mainly around the business work time and > have a small window of safety to not overload the circuits to the point > you will have busy signals often. If it is likely you could grow beyond > 500 people soon, you may want to buy the T400P card and be able to > deploy more digital trunks without taking the system down for more than > an asterisk restart. > > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Traffic Modelling (was IVR only system with scalibility...)
Nice goin'!! I will use this for a reference point to establish baseline numbers of phone lines. The good news is that the equation is not linear (eg 45 people need 5 lines, 100 need 10 lines). So I can double my potential users and ONLY need 2 more lines (qty 7). The bad news is that I don't 100% think I will have purely random connections. On Thu, Sep 04, 2003 at 12:48:58PM -0700, George Pajari wrote: > The question was posed: > > "incomming calls for 45 or so people that will call in 3 or 4 time each > day during (approx) normal business hours" > > The comment was made (taken out of context): > > "The quick math says that 45 people with 4 calls is 180 calls a > day. In a 8 hour day you have 480 minutes. From 480 minutes 1 port could > handle the load if the call was under 2.5 minutes long and everyone > waited till it became available." > > Unfortunately as we all know, asking callers to guess when the line is > free and equally spacing their calls is not terribly realistic (as the > author of the comment above goes on to imply). > > So how does one analyse such a situation? Using statistical traffic > modelling! > > For more information, see http://www.erlang.com/calculator/erlb/ > > Plug in: > Busy Hour Traffic: 0.937 Erlangs > (based on 45 * 4 * 2.5 / 480) > > Acceptable Blocking Factor: 1% > (we will accept 1 in 100 calls receiving a busy signal) > > Result: > you will need 5 incoming lines. > > If you are willing to tolerate (say) 3% of calls receiving a busy > signal, you can get by with 4 lines etc. and etc. > > Hope you find the above useful in planning your Asterisk installation. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Request for consulting time wanted for project
Hello, I posted a question a few days ago and as part of a discussion, someone mentioned asking the list for consulting help...so I am. I originally type something out and sent it to the admin portion of this list by mistake. This is more hopefully correct attempt. If you are interested in providing consulting services, please read on. We are looking to implement an IVR system for user call in. The users's will likely call in 3 - 4 times per day. We are attempting to gather timely delivery information from theses users. The CURRENT documented script has approx 37 key entries. We would look to reduce this number by any means usable. The dial plan would need to link to a database for the common authentication/authorization/verification tasks. We do not expect the project to be difficult for someone familiar with setting up *. We are looking to establish a proof of concept/feasability to head off the need to OUTSOURCE/BLACKBOX this project. We are very good at Linux/perl and are trying to learn as much as we can on the IVR/PBX side as we can. We want to "do this right". I believe that part of that is to know what was done and how it works (eg I want to learn). We are a US based company and have contracted open source developers for our needs in the past. I would look forward to recieving/sending information from interested parties. Please include information on past projects (your scope of the project please), and your per hour cost or if you would want to do a project pricing. Again, for someone with the experience, I do not believe this to be hard. We need help to prove we can internalize this project or loose it to the outsource/blackbox/money people. I do not want to give in to the dark side... Thank you Please email: pj at cassens dot com Please note that I will not be availible today, Sept 10th very much. I will likely only be able to respond beginning on the 11th. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] say number question
I searched for "say number" in the * google archives and have not found reference to options for "say number". I would like to have * say digits instead of the hundreds and thousands. EG, "1234" would say one two three four. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] say number question
Oops should have looked a little harder to find the "say digits". Sorry. On Fri, Sep 12, 2003 at 10:21:32AM -0500, PJ Welsh wrote: > I searched for "say number" in the * google archives and have not found reference to > options for "say number". I would like to have * say digits instead of the hundreds > and thousands. EG, "1234" would say one two three four. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and defunct perl procs
Trying to figure out why I'm having all of my test (and demo) perl script in a defunct status. Each run creates a problem: ps output root 26253 1356 0 16:39 pts/100:00:00 asterisk -vvvc root 26270 26253 0 16:40 pts/100:00:00 [pj.pl ] root 26271 26253 0 16:40 pts/100:00:00 [pj.pl ] root 26273 26253 0 16:40 pts/100:00:00 [pj.pl ] root 26292 26291 0 16:45 pts/100:00:00 [pj.pl ] root 26294 26291 1 16:45 pts/100:00:00 [pj.pl ] root 26295 1259 0 16:45 pts/000:00:00 ps -ef ASTERISK output -- context = default -- dnid = unknown -- enhanced = 0.0 -- extension = 22231 -- language = en -- priority = 1 -- rdnis = unknown -- request = pj.pl -- type = Zap -- uniqueid = 1063403094.0 1. Testing 'sendfile'...PASS (0) -- Playing 'digits/2'. -- Playing 'digits/2' -- Playing 'digits/2' -- Playing 'digits/3' -- Playing 'digits/1' PASS (0) == Complete == -- AGI Script pj.pl completed, returning 0 -- Hungup 'Zap/1-1' the pj.pl is nothing but a hacked up agi-test.agi file. Any help? Mailing list archive shows only a reported problem with festival (search on defunct). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and defunct perl procs
Yes, this is RH9. Thank you for the info. On Fri, Sep 12, 2003 at 02:59:46PM -0700, Scott Stingel wrote: > If you're running RedHat 9, there is a known problem. > > Try executing the following line in the shell before starting asterisk: > > export LD_ASSUME_KERNEL=2.4.1 > > Hope this works! > > -Scott > > Scott M. Stingel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gnophone and RH9 comments and questions
Hello, trying to get gnophone to fully function under RH9 and finally have the browser working from the rpm from 2001. Turns out I had to get an old Mozilla 1.2 version and alter the startup gnophone script to point to it (MOZILLA_FIVE_HOME and LD_LIBRARY_PATH). How much better is the CVS stuff? I cvs checked out the gnophone and the Changelog only shows a comment about 0.2.5 but no real changes. Is that even accurate? Next, has anyone hacked their own web scripts to replace the iaxtel.com page link stuff with more internal fun stuff and willing to share?? Man the memory footpring of gnophone is big. I know, I know it's the browser... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog FXO Card
Looks *exactly* like the X100P (X101P) card I just got from digium last week... On Mon, Sep 15, 2003 at 11:03:14AM -0400, Daryl G. Jurbala wrote: > > > > "These cards are replicas of the X100P sold for use in an Asterisk ( > > www.asterisk.org) phone system. They are fully functional and > > work with the > > same wcfxo driver as the actual X100Ps." > > > > So is someone pirating digium hardware? > > Interesting that it has 2 ports on it, and a speaker. The picture looks > a whole lot like a modem to me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call center design question
Yes, Please share. On Tue, Sep 16, 2003 at 03:05:33PM -0400, Yifang Dai wrote: > On Tue, Sep 16, 2003 at 03:27:44PM -0300, Paulo Mannheimer wrote: > > Hi Rich, > > > > We have done this before. We basically developed a small client that > > sits on every machine and communicates with * to get information about > > an incoming call. Contact me off-list and I will be glad to tell you > > more about the entire solution. > > > > Hi, I'm interested in this solution too, can you share it with the > group? Thanks! > > -- > Yifang Dai | > eFax: (847)628-0255 |Debian GNU/Linux > [EMAIL PROTECTED] | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Source?
You guys are a tough crowd. I do have to admit I did "get" this one, however. I don't know about Senad, but this is not an easy list to pick up on. In order to search the list, you have to know the terms/acronyms. In order to know the terms, you have to learn/ask. Many of you know this stuff back and forth. You know the relaionships of what-does-what. You have connected the dots and put these pieces together. I am still trying to get a handle on MOST all of this stuff. I can barely get the demo to work ;) Let's face it, there will always be dumb questions (like most of mine). Please be nice and think of the many factors that can contribute. Think of knowledge and language and barriers. This */IVR/VOIP/Telephony stuff is only easy when you get to *REALY* know it. I am not there! I know my GNU/Linux systems... I don't know this... please be nice to me atleast ;) On Wed, Sep 17, 2003 at 10:38:58AM +0100, Alastair Maw wrote: > Senad Jordanovic wrote: > > have you more info on this free phone offer? please send it to me off the > > lest? > > Just as a totally wild guess, and call me crazy and amazingly > intelligent for thinking of it, but how about looking at www.nikotel.com? > > I remain astonished by how many people need constant spoon feeding... > > -- > Alastair Maw <[EMAIL PROTECTED]> > MX Telecom - Systems Analyst > http://www.mxtelecom.com > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Source?
I have to defend us newbies on this. This environment does not facilitate sequential knowledge building! Based on my entry to Asterisk, I should have already known T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you get the idea (still trying to figure out "skinny"...cisco something, I know). Heck, I'm struggling to get a grip on what and how to use/confiure SIP for linux and keep my hair. You don't start off with a prerequisite of knowledge to join like a class/school. You don't have the you-must-have-asterisk-101-before going to asterisk-102 before you can join this list. You have a forum that is GENERAL. I would like to a better effort to provide a more sensible way to start helping us newbies. I have to say that the Digium handbook helped a little, but not much. I have googled till I couldn't see straight. I just don't yet have the "big picture" that most of you do. I couldn't even tell you if I need a channel bank or a channel changer ;) at this point. A group of you seem to expect people to have a knowledge base that allows for entering keywords to google. I don't know those keywords. You know the context to search for when someone says I'm having a problem with insert-thing-here. Instead of the usual, "Search the archives". It would be more helpfull to give a hint on what to search for. I could search for SIP and get back several hundred "answers". Then I have to figure out where that answer lies in the series of possible answers. Then I have to somehow figure out if it works. As most of you teachers (past and present) should know, not all of us learn the same. Some people just "get" written material. Some NEED the "spoon" to make it to the next level. Some need the hands-on experience and other's just can't learn any more than they have already know(those people are not likely on this list, however). You do realize that the http://www.asterisk.org/index.php?menu=support lists the mailing list first for support, don't you. In fact, you have to go to the second page before you even see the google reference. More a few people tend to look for the FIRST way to get help not ALL ways to get help... On Thu, Sep 18, 2003 at 08:31:59PM +0200, Dave Cotton wrote: ... > Absolutely agree with you Steve. I left teachers training college in > 1970. I shock some teachers when I said that in all the years since I > haven't taught anyone anything. I've just enabled them to learn. > The problem is that in most national education systems the teacher is > expected to provide the answers to pass some test at the end of the > course. Thinking is not part of the curriculum. > -- > Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Source?
Sorry about changing the original incorrect subject of "Re: [Asterisk-Users] Grandstream Source?" . Many have already written that thread off and this may be a good place to start on a positive note. Yes, I forgot to mention some of the sites that I have found usefull. I do have to say that http://www.voip-forum.org/ has been a very good resource! Keywords: newbie help support "search google" documentation links "spoon feed" So, I would say that these are some sites of interest in no real order: http://www.voip-forum.org/ http://www.asterisk.org/index.php?menu=support http://www.fnords.org/~eric/asterisk/ http://asterisk.gnuinter.net/ http://megaglobal.net/docs/asterisk/html/ http://home.cogeco.ca/~camstuff/ http://www.wwworks-inc.com/asterisk/ http://www.google.com/custom?q=&sa=Google+Search&cof=LW%3A40%3BL%3Ahttp%3A%2F%2Fwww.asterisk.org%2Fimages%2Ftopics%2Fasterisk.png%3BLH%3A40%3B%0D%0AAH%3Acenter%3BGL%3A0%3BS%3Ahttp%3A%2F%2Fwww.AsteriskPBX.org%3BAWFID%3Ad7bc203313616854%3B&domains=www.marko.net&sitesearch=www.marko.net Please feel free to add to this list On Thu, Sep 18, 2003 at 09:53:44PM +0200, Olle E. Johansson wrote: > I realized the same and started a process to collect a lot of that information and > build a > knowledge base on http://www.voip-forum.org/ > > Click on "Asterisk" on the home page and you'll find a lot of information. On that > web, you'll also > find information I gathered about the rest of the telecom stuff I didn't know > anything about. > So have others. There's plenty of pages with facts, explanations and pointers to > find there. > > It's a start, please help us helping other newcomers by adding stuff, questions and > keywords > you don't know. If you haven't got an explanation, create a page named by the term > you > don't now and simply add "What's a pyroflax?" on it. Someone will notice and explain > what a pyroflax is... > > The environment surrounding the Asterisk Open Source project is built by all of us. > Now, you're part of this environment. Welcome! > > /Olle > ...still learning and trying to understand FXO, ISUPs, RDNIS and other terms... > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Newbie delimas was "Re: [Asterisk-Users] Grandstream Source?"
I expect a user list to be for users' questions. I expect a user list to support that what it's a list for. In return *I* should help someone when/if I can! There is no "for Nothing". You help me, then I help some newbie 10 years from now when I understand this stuff. So, in the meantime, my only contribution is the list of sites I have found to be usefull. I forgot to change the subject line, however. I am finding that it's hard to find out what's available when I don't know what's available... Don't get me wrong, I would like for this to be a *constructive* thread! I don't not want anyone to get offendend. I would just like a general realization of the newbie situation. I still think this list is good! I still think * is great. I still have faith that I can figure "all of this" out. I know it will take the help of many good people with ALOT of patience and understanding and experience to help me. I am very greatful for all of the information that you list goers have provided! So many of would do anything to help and do. The more I search through the archives, the more I know that I'm still in the "right place" to help me. Again Thank you for your understanding, help, time and effort! On Thu, Sep 18, 2003 at 04:19:04PM -0400, John Vozza wrote: > BS! :) > > Take the time to read and learn as much as you can from what's available > and believe it or not you may just learn something. Even if that something > is what to ask/search for. > > All those that get paid to answer questions on this list please raise your > hand. I know my hand is still on the keyboard. > > I always amazes me how so many EXPECT so much for nothing... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Source?
On Thu, Sep 18, 2003 at 06:09:27PM -0400, Steve Creel wrote: > I am NOT a VoIP guru. I am NOT an Asterisk guru. I am NOT a telephony > guru. Take that as a disclaimer for the information below, as well as to > say that the best learning comes from reading anything you can get your > hands on. The idea of "post any question to the mailing list" works well > with 10 people. It scales horribly. Reading through the archives, you > will see the same questions asked (and answered) over and over. At _some_ > point, it's okay to say "I've answered it 15 times, YOU can go look it > up on YOUR time". Besides, I'd rather spend 3 hours looking for the > answer than just ask my question, because I hate looking like an idiot. > > This isn't a flame, nor a sarcastic, snide response. I don't want to > complain about people asking "what is a " if I've never made an > attempt to answer that question for someone. > GREAT stuff! Thank you very much. I was very pleased to see that you took time to describe all of the "T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank" stuff I put down. I hope this thread will end up in the hands of a new newbie and can help... Thank you all for helping. I had to say something and I felt this list (the people) could handle my comments. I'm glad to see that I was correct. For my part, I will try to stop top posting and dig alot deaper into the archives. I realy do want to learn this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CHANGE THE SUBJECT LINE Re: [Asterisk-Users] Grandstream Source?
On Thu, Sep 18, 2003 at 09:22:55PM -0600, John Brown wrote: > HI folks, nice conversation, but it has *nothing* to do with > the subject line. Sorry, I tried twice (forgot once and did a second time). Someone else also tried and now you. End result... I think this may be beaten to death... for now... only for now... it will return... More good links to share for the newbies: #Getting Started With Asterisk http://www.automated.it/guidetoasterisk.htm #gota love the name. usefull conf http://www.asstricks.org/ #more example confs and sounds http://www.loligo.com/asterisk/ #dudes bookmarks on VoIP, SIP, H.323, RTP http://www.ict.tuwien.ac.at/darilion/bookmarks.html #Asterisk Guide for INOC-DBA from archives http://www.pch.net/resources/discussion/inoc-dba/archive/2003-July/000719.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
Does this thread help? http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html On Fri, Sep 19, 2003 at 01:18:53PM -0500, Peter Zeltins wrote: > I wonder how IAX compares to SIP bandwidth-wise? I've tried both over > overseas IP connection, and somehow SIP seemed to work better. > > Peter > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P question.
I am about to try our TDM400P E model from the developer kit (not the "Lite") we just got and noticed a large number of reported problems. I had the CVS from Sep 12 (or so the CVS/Entries file has in it). My drivers seem to modprobe fine. My card show up as "Found a Wildcard FXS: Wildcard S400P Prototype (4 modules)" in dmesg and no apparent errors...yeah. Is "S400P Prototype" OK for this card? My real question is physical appearance of the card. ALL the pictures show 4 modules attatched to the TDM400P (on the top edge of the card from front to back). Mine only has 1. So, Should I have 4 modules or 1? What do my missing modules do? Nothing found with the search for "TDM400P" via google. The current google search from the digium does not yet register any of the recent stuff yet. "A search by thread or author option will be available soon!" for the mailing list archives. PS My copy of the install sheets from Digium seem to ommit any reference to "modprobe wcfxs" when you have a TDM400P. I remembered that from previous emails I read. See, I can learn ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + NAT Howto?
Don't know yet if it helps, but if you read the link at: http://www.voip-info.org/tiki-index.php?page=NAT+and+VOIP it will point you to: http://www.sipcenter.com/files/SIPNATtraversal.pdf However has the voip-info.org site; your stuff ROCKS!! On Fri, Sep 19, 2003 at 03:11:31PM -0500, C. Johnson wrote: > Hello Folks- > > Pretty new to the list here, got a lot of reading to do.. Does anyone > know where I can find a decent HOWTO or set of instructions for > running > Asterisk and SIP clients thru firewall/NAT systems? > > I have a Asterisk box sitting behind a linux firewall at a remote > location > and have the 5060 and etc ports open as well at 16381-16391 UDP open > and > routed to the Asterisk box as well. I have a bunch of clients at > another > location which are also sitting behind a Linux ipchains/tables > firewall > > > So far, I'm able to get the clients (Xten Lite) to ring each other, > but they > ring, and one will say it's connected, while the other one just hangs > up. > > > -cj > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MS Outlook
What am I if I use mutt...besides virus free? ;) On Mon, Sep 22, 2003 at 04:30:49PM -0400, Sean Heiney wrote: > Actually, MS Outlook by default blocks all executables. I'm not sure why > there is so much negativity around the Outlook client. Perhaps we > should all go back to the cave and use Pine. > > > -Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Anyone looking for IP Phones?
On Mon, Sep 22, 2003 at 04:33:44PM -0400, Ariel Batista wrote: > > I am very interested in why your no longer using these phones. We have one for > testing and so far it's not working. Are they not working? Is the main problem > configuration? Any information will be very helpful. > There web site indicates that they are ligquidators of dot bomb's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New kid on block
On Tue, Sep 23, 2003 at 08:54:50AM -0400, costas wrote: > Hi, > > I am an experienced developer with Windows and familiar with Linux. I am looking for > a SIP solution. > > 1) How does Asterisk compare to VOCAL in terms of support. Sorry, don't know > > 2) Is Asterisk free? yes > > 3) Where are the docs? Or even better. Where do I start? Take a long read at the sites below. The www.voip-forum.org is VERY useful! Read both of the * Handbooks for asterisk.org. Spend a little time with the google search below with you qutestion. http://www.voip-forum.org/ http://www.asterisk.org/index.php?menu=support http://www.fnords.org/~eric/asterisk/ http://asterisk.gnuinter.net/ http://megaglobal.net/docs/asterisk/html/ http://home.cogeco.ca/~camstuff/ http://www.wwworks-inc.com/asterisk/ http://www.google.com/custom?q=&sa=Google+Search&cof=LW%3A40%3BL%3Ahttp%3A%2F%2Fwww.asterisk.org%2Fimages%2Ftopics%2Fasterisk.png%3BLH%3A40%3B%0D%0AAH%3Acenter%3BGL%3A0%3BS%3Ahttp%3A%2F%2Fwww.AsteriskPBX.org%3BAWFID%3Ad7bc203313616854%3B&domains=www.marko.net&sitesearch=www.marko.net > > 4) Will it run on RH9? My AGI's seem to go defunct on RH9 > > Thanks in advance. > > Costas > > -- > Costas Menico > Meezon Software Corp > 201-224-8111 > [EMAIL PROTECTED] > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie: Crossing my fingers
On Fri, Sep 26, 2003 at 06:40:12AM -0700, TC wrote: > Welcome > I have been updating this doc with links to user documenation > as i come across it > http://bugs.digium.com/bug_view_page.php?bug_id=070 ERROR: Access Denied. as user anonymous... guess I need to create an account. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Web Search Frontend
On Mon, Sep 29, 2003 at 11:09:06AM +0100, WipeOut wrote: > > >I was thinking of using > > > >http://developer.berlios.de/ > > > >As SF has had many problems recently :( > > > >Regards > > > >Mark > > > > > > > Yea, I have noticed Sourceforge has been a little flaky lately.. Thought > they would have been on top of it quicker.. http://developer.berlios.de/ seems to be down for me in the US at this time: * Connection Failed The remote host or network may be down. Please try the request again. Generated Mon, 29 Sep 2003 12:51:36 GMT... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Web Search Frontend
Comments inline: On Tue, Sep 30, 2003 at 07:38:07AM +0100, WipeOut wrote: > Mark Evans wrote: > > >>I think we're getting away from the original purpose of this program. > >>Are people really that desparate for a full, web-based admin/user > >>interface? > >> > >> > > > >I sure am, I want to give as much control as I can for basic tasks > >to my customer who may not even know what Linux is :) > > > >Regards > > > >Mark > > > > > > > So it sounds like we need to combine everyones thoughts into a feature > list of some sort.. > > To me it sounds like we have talked about the following.. > > * PHP based > * Database Independent (Initially MySQL and PGSQL) > * CDR Viewing (real time view of last 20 (filterable by > src/dst/accountcode etc..) CDR entries) > * CDR Reporting (per user/company/line, Somthing like the itemised > billing from a telco, User Accessable) Do you want everyone to know where everyone is calling? I could see mgmt/owners having an issue with this in a normal business environment! Look Mr. Gate$ at extention 666 made 500 calls to Mr Devil last week, I wonder what deals they are making ;) > * Asterisk Management (initially based on PHPconfig) Doesn't this just display the config files for editing? I wouldn't want to put this as an internet reachable service without https. Heck if I'm just editing a file, why not just vi/vim/emacs/your-fav-editor from an ssh session ( lots of java ssh applets for this/ ssh clients )? User Self Service would seem to mean replacing the PHPconfig anyway... > * User Self Service (Ability to change VM password, Maybe even SIP > password, Advanded features in the future could be things like call > forwarding setup or a facility where the user could sit down in front of > any phone and "map" his extension to the phone on the desk in front of > him like hot desks in an office) Make sure to think about limits for call forwarding. Should I be able to forward to a number in some other country? Authentication/Authorization is a tough slippery slope... LDAP, NIS, NIS+, flatfile, database you get the picture. You may want to think about Webadmin as a control frontend and build with that in mind... Maybe CDR exports from the GUI (csv)? In GUI Archiving mechanism for "old" records? ( record retention policy ) > > Feel free to add you ideas.. > > Later.. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Web Search Frontend
On Thu, Oct 02, 2003 at 09:58:37AM +1000, Jamie Carl wrote: ... > As for the rest of this discussion, I have already started > work on this Asterisk Web Interface. (visit > http://astweb.sourceforge.net). The current release is > still only the CDR section, but things are starting to > evolve and I expect to have something usable in the next > few weeks. It is being written in PHP and will attempt to > use ZERO OS-DEPENDANT code. ... Sorry, preview selection generates: Your DNS2Go account has been disabled. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any way to get out of a remote console without stopping *
on Thu, Oct 02, 2003 at 02:53:00PM -0500, Andy Hester wrote: > This probably has an easy solution, but I found it yet. How can I get out > of a remote console after using ssh to get into the box, making changes, > reload etc. without stopping *? > > Thanks in advance. Looks like "exit" will release you from the * console but not stop * from running when I start * with "asterisk -vvvc". Then "asterisk -r" to reconnect. I like to use the "screen" command to preface other console grabing prgs. screen -A -m -d -S asterisk asterisk -vvvc then screen -r asterisk connects you to the screen you just called "-S asterisk". You use key binding similar to minicom to do things... so to release your screen from running * while in a "screen" session (and not kill the *), just "+a" then hit "d" to detatch. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with questions for initial Asterisk wizard (GUI)
On Tue, Oct 07, 2003 at 04:40:36AM -0500, Steven Critchfield wrote: > > What point do you feel that a user is too advanced to us your wizard, or > at what point do you think a user of your wizard will be more pissed at > being hindered by the product than helped? > > I'm not trying to insult you, or necessarily put down what you want to > do. I just feel that it is way to simplistic to think a wizard will make > anyone happy but a small fraction of users. If you > > -- > Steven Critchfield <[EMAIL PROTECTED]> Good suggestions for changing his program. I strongly believe this kind of "coddling" will be helpfull to many more people that you expect. First, a large number of people are coming in with the "I saw * and want to try to do a ??? phone" or "I have a X100P and want to do". Those are realy (according to you old timers) trivial. Most of them are looking for help getting started...here is that start. Many will only need to run through a simple config and can them look at the conf files for more "advanced" setups. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * consultant needed - will pay
Thank you for reading this, sorry to waste bandwidth otherwise. I am part of a US company looking for someone to setup a demo IVR system for us. I seem unable with my current knowledge to pull this off myself. The demo is the regular enter your id and validate/repeat/continue methodoligy you put up with in everyday life. I would like to have the validation and other parts done via database (Postgres or MySQL). This is a FOR PAY job, with the potential for landing the full project. I need a quick turnaround! I have gotten myself in a serious time crunch before I have to go with another proposed M$ solution and a great deal more money. I need for contacts as soon as possible. I would ask that you be able to accept either PayPal or PO or work till you get a check. I DO NOT know how to handle the potential for transactions outside of the US. If you are outside the US and can still accept US $'s and know the implecations, I think we can work something out. I can provide additional info to interested people email to: pj at cassens*dot*com I will reply as soon as possible. Again, I need a quick turnaround! Skills in * + database + AGI are likely manditory. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Project Completed [Files Attached]
I have not found the type of license you are using for this demo. Can you please confirm which one you plan to use for this. On Sun, Oct 19, 2003 at 05:16:29AM -0700, Azher Amin wrote: > > Thnx for the interest in the ivr sample [btw: I am not an expert in PERL/AGI :) ] > comments are welocome. > > You can download the demo files and sounds from > http://www.consulttech.com.pk/asterisk/IVR.rar > > There is a flowchart in the excel format thats shows how it works. > > you will need to place the sound files in the /backup/en directory, or u can change > the code as well. > > I will be putting another application as soon as it will be prepared. > > Azher > > > > > - > Do you Yahoo!? > The New Yahoo! Shopping - with improved product search ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rollout tips
On Tue, Nov 04, 2003 at 02:49:39AM +0100, Christopher Arnold wrote: > On Tue, 4 Nov 2003, Shoval Tom wrote: > > > Olle, www.voip-info.org still resolve to 192.168.168.3 from here, and many > > other places (like our branch office, my home dial-up account, my parents > > dial-up account) > > > Do you by any chance use the same ISP? > Or migth it be so that all accounts use the same dns? I may be usefull to know that I have never had an issue getting to www.voip-info.org from a Verio or a Charter account. I realy think you have made alot of very usefull info availible. Thanks for you effort. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for downlaod...
> From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Dan > Sent: 03 November 2003 19:18 > To: Asterisk Users > Subject: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for > downlaod... > > as promise, at: > > http://www.laser.com/dante > or > http://www.geocities.com/tdanro As an interim option, someone with wine or codeweavers cxoffice knowledge may be able to figure out what it will take to runn on linux. I tried with cxoffice and get errors. I will try a little more with my VERY limited knowledge to work something out... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX/IAX2 encryption?
On Mon, Nov 10, 2003 at 03:26:06PM -0500, Brian J. Schrock wrote: > > I second that, and I think I remember hearing Mark talking about it too. But. > > What type of encryption can you do that does not introduce latency? > > That said, I would like it to support hardware encryption cards. > > I have done work with FreeS/WAN and it works, and yes it adds about 30-100ms of > latency depending on what else is going on. I think it has something to do with > keying. I don't understand why the latency will be so high. I've run misc test (not with * since I don't have a PBX/voicemail needs with *) and find that I have less issues (more consisten responces and good throughput) with FreeS/WAN. The firewall machines maintain a persistant tunnel. They should be keeping "active" connecitons between servers humming right along. Do you have an overloaded FreeS/WAN server? I even get better results going through FreeS/WAN on one connection to my home (the cable provider seems to like to bandwith throttle the other services to some degree). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL Licence may be changing..
On Wed, Nov 12, 2003 at 09:22:56AM -0800, Chris Albertson wrote: ... > There are several other very good free SQL DBMSes. One of them > is actually supported by one of the world's largest software > companies, SAP. > > SAP and MySQL signed an agreement where MySQL will co-market SAPDB > and the name will change toi MaxDB. MaxDB is be marketed as a > "step up" from MySQL to an "enterprize class" DBMS. > It will be interresting to see how the MySQL people will define > MySQL, they surely will not try and tell people it is "enterprize > class". Perhaps they will be truthfull now and sell it as a > light--weight product. At the Portland Open Source Conference, the MySQL guys basically called the SAPdb crap. They were going to take some of the good aspects of it an put it into MySQL. They said the SAPdb was not extensible in the form it is now. Of course, they never actually called it "crap" that is my summary of the conversation that I heard. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Your thoughts..
On Fri, Nov 14, 2003 at 01:08:50PM -0800, Steve Sobol wrote: > Senad Jordanovic wrote: > > > Funny, I am doing the same at the moment... :) > > > > We are allowing * to dump call records onto a remote database server. > > Once there we can do all sort of things with it. > > > > My only concern is if this remote server goes down! > > Use more than one database server... How much money or time do you want to spend to "make sure" it gets there??? You could cluster the database (a couple of ways with and with out the db servers knowing). This could include a "real" cluster and load balancer and redundant whatever. This could also include something like "sqlrelay" http://sqlrelay.sourceforge.net/. You could also add an intermediate step that can be "smart" about the connecting service. This could be as "simple" a cron job that attaches a csv file to be emailed to a program that will process the data (we do it all the time). MTA's have VERY good characteristics for service outage issues. This option could also include a more custom daemon (done this, also) of your own creation of something found on freshmeat.net or sourceforge.net (I can't remember the program names off hand) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Crashed Asterisk
On Wed, Nov 26, 2003 at 01:55:36PM -0500, Clif Jones wrote: > Thanks for the truly useful feedback. I'm having a real hard time with > the FAQ pages listing > RH 8 & 9 FIRST in the list of Linux distros that Asterisk compiles and > runs on and having > any bugs (oh I mean RH problems) discarded. It would be much more help > to have responses > such as yours or to have RH removed from the supported distro lists if > it truly does not work right with > Asterisk. It would also be useful for the experts to tell everyone else > which Linux distros are > supposed to work 100% with Asterisk. > > One more question for the enlightened: How does the Debian Distro > (Woody and greater) play with Asterisk? I have seen odd issues with RH9. The same code works on RH7.3. Some people suggest moving to Debian or Gentoo. I am trying to stick with RH7.3... at least so far. I am using * strictly for IVR functions. YMMV. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PREPAID APPLECATION
It is a shame that within a couple of hours they can tell you to remove helpfull documentation, but not (seemingly) help answer questions regarding there Cisco stuff on this list. I think Cisco must have their priorities mixed up! Just my opinion... which also means I won't support a company like that... so I won't buy their products... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Asterisk overwrite any libraries?
On Wed, Dec 03, 2003 at 10:42:40PM -0500, TeleSIP wrote:
> A good rootkit will also modify the date and time of the replaced binaries
> so they will look the same as the original.
>
> Try to replace your "ps" command with that from a trusted RH9 machine. If
> it works ok then you must do a clean install to get rid of the rootkit.
Using the RPM database for package verification is a good way to check, also (better
than date/time stamp). So:
rpm -V procps
procps is the package for ps and some other commands, "V" = verify the whole package.
This should NOT return ANY error or information. So, if you get something like
"S.5T c /bin/ps" or ANYTHING else for THIS package youv'e got a problem.
This doesn't 100% work on all rpm pkgs. You often modify config files and they show up
like this:
rpm -V ypbind
S.5T c /etc/yp.conf
This means that you need to use some judgement. Generally, if you have a binary, it
should not change. Configs will or can change.
You could also look to do:
rpm -qf `which ps` # this should return a like that says procps-{version}. If the
output of this rpm command shows, "file nohup.out is not owned by any package" you are
running (based on your $PATH variable) the wrong ps command. This only works for rpm
installed pkgs, not your normal tar installs. This is just one of the pluses for a pkg
manager (not just rpm).
These are based on the partial belief that the hackers with rootkits aren't
"upgrading" your procps package to there version. Basically, this is just another clue
to look at and should NOT be done in isolation.
For some better options, check out:
http://freshmeat.net
and search for "system integrity" then "Intrusion Detection"
AIDE (Advanced Intrusion Detection Environment) is a standout in this realm (free
replacement for Tripwire).
>
>
> - Original Message -
> From: "Paul Oster" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, December 03, 2003 10:24 PM
> Subject: Re: [Asterisk-Users] Does Asterisk overwrite any libraries?
>
>
> > Looks like your box has been compromised. Try
> >
> > ls -l `which ps`
> >
> > You'll probably find an inapropriate date. Whenever I've diagnosed
> > problems like this, I've found badly installed rootkits.
> >
> > To address this on my production machines, I'm going to insruct the
> > router to only allow traffic that is coming from trusted locations
> > to connect to the box anyplace.
> >
> > I really hope I'm wrong about this Costas, but you should probably start
> > verifying your binaries.
> >
> > If your machine has been compromised, a clean install, and patch with
> > all the updated RPMS is a recommended soloution.
> >
> > Paul
> > costas wrote:
> >
> > >I am using a brand new RH9.0 installation. I installed Asterisk
> afterwards so I am not sure if Asterisk caused the problem below. The ps
> doesn't work. It could also be something else. I also tried installing a
> some video package. But I thought to ask here first if someone has seen this
> before.
> > >
> > >[EMAIL PROTECTED] asterisk]# ps
> > >ps: error while loading shared libraries: libproc.so.2.0.6: cannot open
> shared object file: No such file or directory
> > >
> > >[EMAIL PROTECTED] asterisk]# which ps
> > >/bin/ps
> > >
> > >Thanks
> > >Costas
> > >
> > >--
> > >Costas Menico
> > >Meezon Software Corp
> > >201-224-8111
> > >[EMAIL PROTECTED]
> > >
> > >--
> > >___
> > >Asterisk-Users mailing list
> > >[EMAIL PROTECTED]
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > >
> > >
> >
> >
> >
> >
> >
> > Free 20MB Web Site Hosting and Personalized E-mail Service!
> > Get It Now At Doteasy.com http://www.doteasy.com/et/
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some success with linux 2.6 and wcfxo
On Sat, Dec 06, 2003 at 11:14:40PM +, Tristan 'Minty' Colgate wrote: > Hi, > > My apologies for those on the channel who may take offense to this, atleast > the ones to whom it was not aimed, but the fact is that after making a simple > enquiry on the IRC channel I was in absolute shock... ... Thank you for giving the info on you your trials on 2.6. I am sorry for you experience. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
