Re: [Asterisk-Users] Nested MySQL Commands
Send a query over to the MySql server with only the required parameters and have it do all the processing for you and only returns the results. THe above describes the whole point of a SQL DBMS Server. WHat else could you ask one to do for you? Back to the original question: Can you use the result of one query to feed another without going to the DBMS twice? Yes, Of course SQL allows this. In fact Data base experts would be horrified to see any code that querries a value, holds that value and then querries based on that value. Doing this without holding a lock on the relevent tables is just plain wrong and will result in bugs. You should _always_ write the SQL querry such that only one querry gives you the results you need. Manual Locks are not good, they can be the source of very serious performance problems. For more specific advice you would need to post the details of what you are tring to do and a bit of the SQL you are using. I know that is a nice feature od Microsoft Sql. But have not had a chance to read up on the performance of the new version of MySql... Basically MySQL is very fast when the load is light but scales very poorly with either higher loads or higher conplexity querries. It is good for flat file like problems. The larger DBMSes are slower initially but scale better. (Please CC any replies to my direct email) Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nested MySQL Commands
Under Linux (and other OSes) It's not as bad as that. Even with 128 Perl processes running there is only one copy of the Perl interpeter in memory. Each of the 128 running processes would have it's own copy of only it's data segments. With Perl already in memory the biggest system overhead would be process creation. The best design is the one that minimizes the number of process that the kernel has to create. Notice that this is why the Apache Perl modual is so much faster than using Perl from a CGI script Perl connecting to a central DBMS server is already the model you describe: 128 light weight procees connected to one big process which is the DBMS server. Performance gains will come from writting the SQL so that there is only one transaction and using indexes on the right columns in the database. If you really do have 128 process runing and each one needs to access a DBMS server, I'd say you are going to need a very powerful DBMS system but likly the call volume is not neraly like that --- John Daragon [EMAIL PROTECTED] wrote: Douglas Garstang wrote: Peter, I assume you mean something like this in extensions.conf: exten = _X.,1,AGI(master-dial-logic.pl) and then there's only one call. All logic would be performed by the perl script. This has many advantages. One disadvantage however is that potentially, there could be 120 simultaneous instances of this script running (one per call). Yes, but if you need it to scale efficiently, each of these could be a very lightweight process. If you used each of these to communicate via RPC or shared memory to a process with a small and configurable pool of database connections (which isn't that difficult), you can build a simple and scalable solution. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 removal
Almost certainly a memory leak in mpg123 is not the cause of a system crash. First off, there is no such leak. Second even if there were one the mpg123 process is not long lived. A new one is started for each MOH session. If mpg123 is causing the crash then it may be due simply because of all the extra CPU time required to transcode the music. If there are 10 lines on hold then there woud be 10 copies of mpg123 running on the server. As per the suggestion below it is smarter to transcode the music file ONCE and save it. sox is a pretty good transcoder. Also, be carful what music files you use for MOH. Rebroadcasting copywrited music is not lega;. If the wrong person happens to call, be put on hold and hears your unlicensed MOH you could get in trouble. People who work in the entertainment industry tend to be sensitive to this issue. Use some crative commons works and you will be OK. --- Kevin Bockman [EMAIL PROTECTED] wrote: Chris Mason (Lists) wrote: When I configured this server, I did not do the make mpg123 option. Months later, I read about it and did it, as the client was asking about MOH. About a week later the server crashed, which it never has before. I believe mpg123 have a memory leak. What's the best way to remove it, and is there an alternative that is stable? rm /usr/bin/mpg123 or something like that There are a few solutions to MOH: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it If you are using 1.2, I would use native (codec, not MP3). There should be an example in the sample config file in /usr/src/asterisk/configs/musiconhold.conf.sample - I don't see it on the Wiki. It should be there, somewhere. Must be buried. For this option, you will need to have the sound files in .ul, .gsm, or whatever codec you use mostly. I only allow ulaw, so all of my MOH files are .ul. This way it doesn't have to transcode at all. Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stay away from Grandstream!
Maybe a better way to say it is Know the limitations of the GS phones and don't try and use them outside of those limits. Don't buy ANY phone you've not tested and used yourself for use by a client. My GS phone has worked fine for years. Even if it were to fail and had to be replaced buying two is still cheaper then one of some of the others. The trick is to use them (or anyhting else) only when you know it will work. That said, the GS 100 is not the best thing to put on a receptionist's desk. I've actually had pretty good luck, even getting to exchangeemail one of thier engineers. --- Elene Kinsky [EMAIL PROTECTED] wrote: We have 2 GXP-2000 dead during automatic firmware upgrade. Devices now send out only one ARP packet for default gateway resolution during boot and nothing more! We've contact Grandstream support, but they cannot help. Now we want to send devices to Grandstream for repair but they on longer reply mail! GXP-2000 was very buggy on attended call transfer, and the problem resolved only after upgrading using latest firmware. Overall GXP is OK, but customer support is terrible. Stay away from them! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival questions
I was working on something like this. There are a few isues: (1) First off yes you have to have access to the tesxt of the e-mail. Notice the word test we don't want html or MS word atachments. (2) Next look below nice al the silly junk like quotes are indicated by marks. This has to be converted. The the ascii art in the sig line. OK this is not hard just a whole bunch of Perl scripting or if you are really nuts like me try lex/yaac to define an e-mail grammer There is much to be done here basically we are building a script that any reader (human or machine) would be able to read into a telephone. The BEST format to use a voice markup language not plain text. Festival can read the markup language (3) Finally you spimply scrip it and run it on as command line application on _many_ test emails and listen. (4) assuming #3 above is done integrating it into * is very easy. One interrresting idea would be to automaticaly drop email into voice boxes oruswer could us a menu tree If anyone seriouly wants to work on the above isues please CC me directly about it. Im my opinion most of the work is in #2 which would be implemented outside of the * code base as a stand alone application. --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jul 13, 2005 at 03:47:53PM -0400, [EMAIL PROTECTED] wrote: Hi, Is it possible to setup an Asterisk system that can allow someone to dial in using a DID and listen to their e-mail? Has anyone done this? It seems that basically yes, but quite depends on your local settings. For instance, is Asterisk allowed to read users' mails? As for a user interface, consider the one of the original berkeley mail. It was designed to work in very simple terminals. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Backup for linux/asterisk
The standard UNIX backup program is called dump. Try reading the dump manpage by typing man dump at the shell's prompt. This program has been in common use since maybe the late 80's Quoted from the man page: Dump examines files on an ext2/3 filesystem and determines which files need to be backed up. These files are copied to the given disk, tape or other storage medium for safe keeping Many sysadmins will run dump nightly from a crontab entry --- Steve Prior [EMAIL PROTECTED] wrote: Jeff Glassman wrote: My question is as follows. Is there a backup program that will save to a tape drive or a USB CD Writer so if I mess up an install I dont have to go through a complete reinstall? I saw a few programs out there but they required X windows and from what I read it is suggested that X windows not be installed on an Asterisk box. I recently used G4U from: http://www.feyrer.de/g4u/ See if it does what you need. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Best DB
What is the best truck? A recent survey finds that there are far more Ford Rangr pickup trucks on the road then there are Frightliner 18 wheelers In another survey we find that Chevy outnumbers Porche. Closer to home in the computer world, more people use MS Windows than Solaris. I think Budwieser outsells every other beer. In most organizations followers outnumber the leaders The poor will always outnumber the rich. Still interrested in that database poll? What's the best DB. First you must define best. After you do that the answer is easy. --- David Brodbeck [EMAIL PROTECTED] wrote: -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Top Deployed Databases poll shows following databases in use: SQL Server with 78%, Oracle - 55%, MySQL - 33% and PostgreSQL - 8%. I see they created this with Mysql, 78 + 55 + 44 + 8 = 185% I'm sure if you add in the others we would get to something around 300% deployment. Presumably some sites had more than one type of database in use. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients
--- TC [EMAIL PROTECTED] wrote: There are Ham Radio operators who have used * as a repeater controller and you might try to contact some of them as they will have direct experience doing what you are trying to do. I'd like to know too. WHo's done it. I've wanted to use * as a voip radio link but not gotten around to it yet. If you'd like towork on this let me know, but please NOT on this list, or at least CC me directly. I'll miss any reply posted here. Of course the biggest issue will be simplex vs. duplex. the radios are simplex As for connecting the radios to the computer there are comercial made interface boxes or you can make your own for a few bucks. where do these guys hang out any irc channel or mail list = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients
THere are a number of VOIP links used by ham radio. IRLP has to be the most popular. Then there is Echolink, Wires and some others. My plan was to wrte an Asterisk channel driver for each of these Asterisk then could provide inter-system bridging between the various ham VOIP networks, the PSTN and VOIP Telepony. --- Mark Phillips [EMAIL PROTECTED] wrote: Aha, I see where you're going with this. Firstly, why does it have to be SIP? Are you expecting to be able to have users pick up the phone and dial a radio? If not then there are loads of VOIP for radio apps out there. Many run under linux. All use sound card and serial port. Take a look at eqso.org They have a solution that is free and hooks you up to a load of other users using whatever radio you choose to use. Mark, KC2ENI Glenn Powers wrote: TC wrote: Any one know of software that allows 2-way radios as VoIP(SIP) clients, besides dingotel's usb mic cable trick ? http://www.dingotel.com/2way/requirements2way.asp They might be ok if the SIP client was not hardcode to their own SIP proxy Has anyone tried any hacks to get the 2-way radio SIP client to regsiter to a * box. hmm chan_frsgmfrs anyone? using the usb/mic cable under linux :) *The Asterisk http://www.asteriskpbx.org app_rpt project The integration of 2-way radio systems and reasonable telephony *http://www.zapatatelephony.org/app_rpt.html cheers, glenn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients
Mark, You are getting close to doing what I was thinking of. Let's say your repeater system works with sub-watt power transmitter and a little 1/4 wave antenna. You could get about a mile of range with a shirt pocket sized HT. OK so for that's easy. Next you get 50 of your freinds to build systems like that. then you connect all the asterisk boxes with AIX. What you've have then is a cell phone system running on VHF. If you had an LDAP server that culd be kept up to date with what call sign is in the range of which cell you could make worldwide call sign to call sign calls. Also, with low power xmiter lots of people could afford ot git into it. THere are more things one could do with Asterisk and radio then one person could ever cover. I've been wanting to start up some kind of interest group --- Mark Phillips [EMAIL PROTECTED] wrote: What exactly are you trying to achieve? On my repeater system, I use the RC210 repeater controller with the Phone Patch option. This is then connected to my Cisco ATA and then onto *. Users can initiate phone calls and callers can either command the controller or initiate calls to the radio operators (although I have this bit disabled for legal reasons). There are many Phone patch type devices that could be paired with a radio (MURS/FRS/GMRS/HAM/etc) which in turn would require connection to an ATA. As for where do they hang out. On the radio of course! With masses of specturm available to even the most modest licencee why would we join a IRC channel? You can find me on 53.81MHZ, minus shift, 136.5HZ PL Mark, KC2ENI Michael B. Murdock wrote: Not sure where they hang out you might look here.. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Rpt and here.. http://zapatatelephony.org/app_rpt.html should be some contacts on these pages to get you started. - Original Message - From: TC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 24, 2005 1:38 PM Subject: Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients There are Ham Radio operators who have used * as a repeater controller and you might try to contact some of them as they will have direct experience doing what you are trying to do. where do these guys hang out any irc channel or mail list ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients
THat is a valid conern if there is a path from the PSTN to a ham licenced transmitter then any unlicenced person could get on the air. Isues like that could be worked. Software even enforce any rule such as requiring operators present at control points or PSTN disconnect before IRLP connect You could build a system that talks to multiple services but never allows them to be connected in real time --- Mark Phillips [EMAIL PROTECTED] wrote: I got into SERIOUS trouble with the IRLP folks for trying to do this. They want a closed netowrk and won't entertain anything that could allow a non licenced ham from using their system. Mark Chris Albertson wrote: THere are a number of VOIP links used by ham radio. IRLP has to be the most popular. Then there is Echolink, Wires and some others. My plan was to wrte an Asterisk channel driver for each of these Asterisk then could provide inter-system bridging between the various ham VOIP networks, the PSTN and VOIP Telepony. --- Mark Phillips [EMAIL PROTECTED] wrote: Aha, I see where you're going with this. Firstly, why does it have to be SIP? Are you expecting to be able to have users pick up the phone and dial a radio? If not then there are loads of VOIP for radio apps out there. Many run under linux. All use sound card and serial port. Take a look at eqso.org They have a solution that is free and hooks you up to a load of other users using whatever radio you choose to use. Mark, KC2ENI Glenn Powers wrote: TC wrote: Any one know of software that allows 2-way radios as VoIP(SIP) clients, besides dingotel's usb mic cable trick ? http://www.dingotel.com/2way/requirements2way.asp They might be ok if the SIP client was not hardcode to their own SIP proxy Has anyone tried any hacks to get the 2-way radio SIP client to regsiter to a * box. hmm chan_frsgmfrs anyone? using the usb/mic cable under linux :) *The Asterisk http://www.asteriskpbx.org app_rpt project The integration of 2-way radio systems and reasonable telephony *http://www.zapatatelephony.org/app_rpt.html cheers, glenn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Sports - Sign up for Fantasy Baseball. http://baseball.fantasysports.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sphinx
In a production environment, I would not attempt to run Sphinx on the same computer as Asterisk A few users interacting with Sphinx could consume all of the server's resources and then some. Same goes for DMBS servers, One big N-way join could tie up a CPU for tens of seconds. --- Mark Kidd [EMAIL PROTECTED] wrote: Has anybody managed to implement Sphinx in their * system reasonably painlessly. if so: does it cause any problems with normal * operations. does it place any sort of constant heavy load on the machine. are there options for simple vs advanced implementations. all i am looking for is basicaly for a person to say a branch name. ie: johannesburg thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] speech recognition V 2.0
Sphinx and Festival are good projects. The last I worked with sphinx I was told that it would need modifications to make it more grammar aware, but that was 2 years ago and things may have improved. If not then Sphinx people please let me know when you will add grammars natively or refer me a grammar based engine. Sphinx and Festival are in fact the current state of the art. you are not likely to find anything better. Sphinx can return a probibility network. You can then attempt to parse paths through the network and use the first path (searching in probibillity order) that parses correctly. You can use a LEX/YACC parser and do well enough. (Get the O'Reilly LEX/YACC book. It's easy to use.) I'm impressed with YACC's performance. I have an application with hundres of grammar rules that runs as fast as UNIX's wc utility. Users _can_ learn the subset of grammer. Remember the game zork or the other text based adventure games? People caught on to the limited subset of English. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: load balancing 20 asterisk servers
DNS based load ballancing has it's place, as dose using an application level switch. Say an earthquake takes out your California data center. Shortly the DNS servers will notice and pull that center's record. However do to caches and all this is not fast and users will notice. What the switch does is route at the protocol level between local machines. You can take a machine off line and no one will notice. Works great until the big quake a backhoe takes out a fiber cable ro there is a fire flood or who knows what. protocol level switches have to know about the protocol. You can buy them that work with HTTP, HTTPS and the common ones but I wonder aboit SIP? Getting the RPT to the right * server would be hard beetrer to have a proxy tell the user which * server to go to and nothave to route RTP. --- Miguel Ruiz Velasco Sobrino [EMAIL PROTECTED] wrote: --- [EMAIL PROTECTED] wrote: The DNS approach does not handle single or multiple system failures, only very elementary load balancing over a lengthy period of time. Are you shure of that? I'm aware that the load criteria is trickier, but very possible. If you use DDNS (dynamic DNS) using Bind 9. You have to run a health monitor (like a tcp or ping monitor) in one server (like the dns one) if one server dies, a script removes it's A record automagically from the pool, and even with a script that monitors the load you can dynamically add and remove the entry of each individual server in the DNS server. And you would not need to care if the load balancer sends the SIP stream to one server and the RTP stream to an other or in the case of outgoing connections or whatever extrange situation. Use nsupdate utility for doing DDNS, it's really simple and incredibly powerfull. Also because all the requests are digitally signed, you will likely don't have security problems. You may want to consider a simpler aproach, why don't you balance the load via DNS? If you put in a zone file various A records for the same machine, but with different IP's, BIND will catch the trick and send a different IP (from the pool yo defined) each time a DNS request arrives. That's a simple way of doing that, it will definively work for termination, but you may have to think more who to cope with origiation (outgoing calls), since different clients will be connected to different servers. --- [EMAIL PROTECTED] wrote: We use it on our web and mail server to load ballance across multiple hosts. The way we have it configured it will maintain a session for 15 minutes between a client and a specific server. So long as you have qualify=yes in your configuration files, each client will continue to talk to the one server until they are turned off/ deactivated for at least 15 minutes (or whatever time period you configure into it). I've not tested LVS with Asterisk, but it may be the right direction for you to take. Cheers, -Shaun Matthew Boehm wrote: I've read several other emails and pages on the wiki but none give any deffinate answers. if you have 20 asterisk servers each with 4 pri's, all running RealTime Extensions and RealTime SIPBuddies from the same MySQL server, what prevents you from putting all 20 servers behind a single load balancer? That way all of your UA's can use the same IP to register to; vs maintaining which customer is assigned to which machine. perhaps its just that i am not that familiar with load balancers. i was under the impression that a load balancer could/would send each recieved packet to a different server. this doesn't matter in the case of register requests since all asterisk boxes share same SIP registry database. but what about invite requests and the rtp stream? you would have a majorly broken conversation if each packet in the rtp stream went to a different asterisk box. or are load balancers SIP aware? or is there some sort of session control that the balancer is aware of and will send all packets in a sip session to the same asterisk box? and then what about meet me conferences? if 10 UA's all dial a conference DID number and all 10 get balanced to 10 different servers then they are all sitting in seperate rooms right? hints, opinions, facts...all welcome and appreciated. -Matthew __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK
Re: [Asterisk-Users] UPS for Asterisk
--- Remco Barende [EMAIL PROTECTED] wrote: On Wed, 26 Jan 2005, Michael 'Moose' Dinn wrote: If you need a rocksolid solution have a look at astlinux that can boot * from a compact flash card in read only mode which makes it very hard to break :) You should be able to boot Asterisk using slackware as a base from a 64M CF card or even from a 64M bootable USB memory key. If you use ReiserFS or something similar for the drive that stores all your voicemail, etc then it should come back without a problem as well. Indeed, I'm thinking of using 2 CompactFlash ATA disks. One fully read only with just a small partition writable that will keep /etc/asterisk (astlinux mounts read-only always and only mounts read-write if you need to change/save the config). No worries about unclean shutdown. The second disk I will use for voicemail, and I can swap it every year before it wears down. Better than that, mirror the disk. Then when one drive fails Linux will automatically use the other disk. You can go one step more and define a third disk as a hot spare then after Linux detects the drive failure and switches to the surviving twin it will also bring up the hot spare and begin building a replacement for the dead twin. You can then swap out the dead drive with no need to power down the server and declare the new drive as the new hot spare I would mirror the read-only patition also It you truely want 5 nines you have to set things up so that you can do normal maintanance (swapping out drives, power supplies and the like without powering down. It's probably possible to do it with another distro too but astlinux is already pretty much finished :) And the cost of 512 Mb or 1 GB ATA flash is not much more than a McDonalds meal anyways (1GB is about USD 100 now) For voicemail I could also use a microdrive but I'm not sure what will happen if it breaks and * tries to read/write from it. If that would bring the box down it's no solution. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS for Asterisk
The usual setup for a computer is hosting a critical functions is to use a server that has two (or more) power supplies with an A/C power cord comming from each. You then connect each cord to it's own UPS. I typical small PC server would have two internal power suppies and two UPSes. With this kind of setup even the cruddy UPS describbed below would be just an annoyance and the system would not go down. --- David Brodbeck [EMAIL PROTECTED] wrote: -Original Message- From: Jon Radon [mailto:[EMAIL PROTECTED] I've had good luck with CyberPower, what was your issue? I had two of them. The first one, after about a year, would just randomly switch off or glitch, causing the computer connected to it to reboot. The second one lasted two or three years, then suddenly started acting like the incoming power was off, even when it wasn't. It did this briefly, intermittently for a couple of months, and then the condition became permanent and it would no longer switch to the AC line or charge its batteries. I gave up on the brand at that point, figuring an unreliable UPS was even worse than none at all. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Interface to propriotary system and GPL
If everything on the Aterisk side of the socket is GPL'd I think you are OK. Look at the example of a Linux system running Netscape browing the web when there is a Microsoft HTTP server. Then we have a GPL's system conected to a closed comercial system over a socket (port 80). The grey area is when you link GPL's code to non GPL I think the case of a socket interface is pretty clear. --- Shahed [EMAIL PROTECTED] wrote: Hi All, I am wondering if I will be breaking the GPL, if I write for example, a channel driver or make some modifications to the astrisk source code, to interface at RUN TIME, through sockets, with a proprietary system. Eg. 1. I write chan_xxx + modify asterisk source (make changes + new code publicly available) 2. chan_xxx supports hardware by XXX Corp. 3, XXX Corps interface is proprietary. 4. I write a layer over XXX Corps API, that uses sockets, with the ONLY intent to BYPASS the GPL restrictions (If what I think about them are correct) 5. Asterisk now interfaces at runtime with XXX, but no library linking. 6. I sell the system, make all modifications available under GPL, but don't purchase any sort of license from Digium. I looked at http://www.netrino.com/Articles/LinuxLaw/ and some ML posts, and it seems that perhaps I may be somewhat correct ? Please don't ask we why I would want to do this, because this is a hypothetical situation. I just want to clarify this, for maybe, future use. Thanks Shahed ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual NAT for SIP
Plug the Linux PC directly into the DSL modem and let it run Asterisk, firewall and routing and NAT and your wireless. You will need two network interfaces on the linux box Outside users just will not be able to get in otherwise. --- Serge Schumacher [EMAIL PROTECTED] wrote: Hi, My installation at home use two NAT translations before it reaches the linux box where Asterisk is running on. I use DSL with a Wireless router which fwd all packets to an Windows 2003 box an this windows box it NATing the UDP and RTC packets to my linux box. If I try to connect to it from outside I get this error : Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 4f106a72453f654a for seqno 1 (Non-critical Response) linux*CLI Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 4f106a72453f654a for seqno 1 (Non-critical Response) No such command 'Nov' (type 'help' for help) Nov 30 22:19:17 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 4f106a72453f654a for seqno 1 (Non-critical Response) Well the users can connect but unable to establish a voice call between two SIP clients. Someone a clue how it can be solved or... ? Regs, Serge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk newsgrup proposal or phpBB forum
Hey guys, Look at this example www.scubaboard.com It has nothing to do with Asterisk but is an example of a _very_ high volume meaasge board that _works_. It runs on free software, has low admin overhead. It allows moderators on a per subject level and optional authentication. I have used this system both as a user and I have installed and tested the software on my system. It is easy from both points of view. Built in search, and even e-mail subscriptions to threads. Not much more to ask for. This list is actually becomming usless do to it's success. I doubt I would notice a reply to this posting unless you CC me off line and remove the [Asterisk-Users] lable so it doe not get piled in with 400 other messages. I suggest that people here actually try out scubaboard even if you have no interrest in diving. Check out the photos, for sale ads, technical topics, regional/geographic forums and how it all is kept straight and the search feature actually works to find old threads. And did I mentione-mail subscriptions to threads --- Steven Critchfield [EMAIL PROTECTED] wrote: On Tue, 2004-11-30 at 15:52 -0600, Joe Greco wrote: Not everyone has decent access to NNTP either due to firewalls coporate or otherwise That's why many news servers allow access on alternate ports. :-) On a proper network firewall, it is deny all, allow these few ports. So unless you are running NNTP on a port like 80 or 443, it probably will be blocked. Even then a good admin would have a proxy in place to help cut down the bandwidth usage and would therefore break NNTP. or are under a quota due to the amount of illegal activity that appears there. Add to it the inability to control spam or kick an unruley users if the need arises. NNTP doesn't solve any problems, and phpBB creates a bunch. Better question is why do you feel there needs to be a change? You've missed some best of breed options. A newsgroup by itself may or may not be useful. However, either way, USENET (which isn't entirely limited to NNTP, incidentally) has a bunch of powerful clients that are designed from the ground up for participating in large threaded discussions. This is a major failing of many mail clients. I find it easier to follow large discussions with the text-based trn newsreader than with any graphical mail client I've seen to date - bar none - and trn is old technology. Just the thread tree view itself is so useful, not to mention one-key cruising through the tree nodes. Who said mail needs to be graphical? I know a great many people still using mutt for their mail and it probably will resemble trn close enough for your taste. Of course there are plenty of graphical email readers that support threaded views. I happen to use evolution with threads turned on and enjoy it. Your right, threaded trees are great. I love it when there is enough people using correct enough software to help keep the information correct. Of course we get to the same problem here that not all software mail or nntp actually puts the in-reply-to or references headers in to make the tree view work. Many sites gateway various mailing lists into local hierarchies, for the explicit purpose of solving some of the problems that NNTP doesn't solve, because the medium was designed to deal with the functional equivalent of mailing list traffic from day one. Gateway mailing lists to local hierarchies to solve problems that hierarchies doesn't solve? Sounds like broken hacks to me. Maybe in your rush through that sentence your meaning didn't get fully expressed. As for the design, like many older technologies, NNTP was designed before the unrulely behavior of spammers. While I know there are some private nntp servers that enable authentication to protect themselves, it isn't the norm. You can avoid some of the problems of public newsgroups by making it a one- way gateway, with moderator pointing back at the original list, therefore subject to all the normal list posting controls. And a limit on what a moderator will be able to handle unless it is a program, and then it wouldn't take a moment to get past it. Not that email is any more secure. Setting up a one-way gateway isn't too difficult. Is there interest? I can certainly start one. We already do all the FreeBSD lists and a bunch of other stuff here. I belive there has already been one with URL posted in this thread. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK
Re: [Asterisk-Users] Echo -when software doesn't cut it.
--- [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Kindly post it back here when you find something, IF you find something, as Ive been fighting the exact same problem here, and followed the same route you guys did. Ive given up at this point. At this juncture, Im convinced the telco is sending it out intentionallyor space aliens are bringing it...its only on the PSTN lines...Versleazon in my casebecause if this problem isn't curable, then asterisk is dead in the water as far as Im concerned. I simply can not use a system that sounds like people talking are in a tunnel. At this point, this is the only point of failure that kills the whole idea of using this as a real switch. Now, as far as echo cancellation, Ive looked as some of the docs, like the motorola paper, http://e-www.motorola.com/files/dsp/doc/white_paper/PTECANWP.pdf and some purely theoretical stuff, but their math is beyond me. Unless someone has developed a whizz-bang improvement over the stuff I read, the point of transiting from a two wire to a 4 wire model will always cause echo, period, done, abandon the idea, lets go do something else. The limiting factor I see, is our software echo handling doesn't have the computational horsepower attached (no cpu, or stealing cpu cycles from main cpu) to be effective. Does this mean another piece of expensive hardware? Dedicated proprietary chips like the Moto? I'm thinking so. SO, This is it, critical point of failure...cant go on, at least I cant use it. Yes, * does a lot of other stuff just fine, but this is the immovable object, at least for me.. This is something it must do and do well or its over. Ive been holding my breath, hoping something pops up, but it hasn't. At 09:21 7/1/2004, you wrote: Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none acceptable solutions in my situation. Playing with rx/tx gain values just seemed to quiet the voice down and along with that the echo happened to be less noticeable. I could almost get the echo to disappear with a low enough rx/tx gain, but then the voice could barely be heard, or DTMF tones stopped working. So whats the next step? I only get echo when dialing over the PSTN. Using Nufone to dial a PSTN number results in absolutely zero echo. Do I put in a request for a Telco technician to come out and take a look at the lines? One page on the Wiki says: Most of the telco's have technicians with the equipment necessary to help find the problem if the problem really is their outside plant. However, getting to that person can be a real challenge. Any suggestions on ways to overcome the challenge of getting the right technician on the phone? Mike, Contact me off list and let's see if we can isolate the issue. Can't tell from the words you've used what steps you've gone through to date. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Network Sniffing Calls for recording
Many people use ethereal to capture network packets. I've used it to debug SIP sessions. www.ethereal.com/ In theory one could re-contruct a phone converstion from logged packets but it might take some effort and you'd need to be pretty smart to find the packets from a call from Joe early last week in the morning some time. --- Nik Martin [EMAIL PROTECTED] wrote: The WIKI is your friend: http://www.voip-info.org/wiki-Asterisk+record+calls -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank smith Sent: Monday, June 07, 2004 5:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Network Sniffing Calls for recording how can you record calls with asterisk? I didn't even know this was possible can some one point me to a url for info on this? - Original Message - From: lists [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 07, 2004 1:30 PM Subject: [Asterisk-Users] Network Sniffing Calls for recording Ok assuming I don't want to record calls using * but instead want a dedicated server that listens to a mirror port and records calls. Is there a cheap software package out there for doing this for mgcp/sccp? I know if evern cut over to * there is a way but I doubt I will even cut 100% over to * so I was wonder what the list has heard of for call recording via sniffing my gates. I know there are some out there but $100k for 40 users is to high for my blood. Offlist is fine for all flames and answers since this is a bit off topic [EMAIL PROTECTED] OK it's a Monday when it takes 5 tries to get a email to the right list from the right account. Either that or someone switched the coffee pot to decaf again. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on Compact PCI platform
It can from an NFS mounted image just fine. You don't need to spend the bucks on flash RAM. Linux can boot off the net then do the moont. This machine is also headless (no CRT or keyboard) and no CD, HD or Floppy. I wish the AMD CPU would run without a fan on the heat sink but when I disable the fans temps go way up. --- Jay Milk [EMAIL PROTECTED] wrote: Since this is related... Does anyone have Asterisk working on a Flash-drive? I was considering this as an alternative to having a harddrive in my machine, thus keeping down noise and heat. A 512MB CF card should be plenty to get Linux and * booted, another 64 or 128MB card should be plenty for voice-mail and such. Any takers? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gelson Dias Santos Sent: Wednesday, May 19, 2004 12:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk on Compact PCI platform David H Hickman wrote: I have it working on an industrial single board pc. :) Could you post some more info about your setup? Like board brand/model, what kind of interfaces are you using and even some photos :-) Seems a very interesting project... is there anybody else running a small/compact asterisk system? I would love to have such a small system that I could send to parents, instruct them to turn it on and plug their pstn line and broadband connection and have a pstn x sip intelligent call router that requires no user intervention. Gelson David Hickman TSG Computer Consulting - Auctions 314-865-4752 x2 On May 18, 2004, at 8:42 PM, Jacques Leisy wrote: Anybody running * on a compact PCI platform? I got a few CPCI boards on eBay including a T1 Natural Microsystems AG4000? Any hope to ever get * running on that platform? Linux Suse 9.0 is running fine Thanks Jacques ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Domains Claim yours for only $14.70/year http://smallbusiness.promotions.yahoo.com/offer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreeBSD
One other on-line resource: http://zapatatelephony.org/ The _original_ driver was for BSD. The folks at Digium appearently ported it to Linux and likely improved it over time. But the old BSD driver is still there. The zapata cards were a kind of open source hardware project. The hardware design was made public See above URL. Getting the old BSD driver to build would be a good first step. See here for old BSD driver http://zapatatelephony.org/tor.c --- Jon Myers [EMAIL PROTECTED] wrote: head on over to http://people.freebsd.org/~blackend/doc/en_US.ISO8859-1/books/arch-handbook/index.html and have a look at chapter 13, Writing FreeBSD Device Drivers, and have fun. =) If I didn't have all sorts of other half finished projects, I'd mess with it... but I've only done simple apps before, nothing like writing a driver.. fobbit.org has gotten the old Creative VoIP blaster working under FreeBSD, but its a whole application, and not just a driver. - - - Jon At 09:54 PM 3/29/2004 -0400, you wrote: *cough* not a thing. - Joshua Colp. - Original Message - From: James Moran [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 29, 2004 9:46 PM Subject: RE: [Asterisk-Users] FreeBSD Does any of the hardware work with FreeBSD?? On Mon, 2004-03-29 at 20:25, Steven M. Sokol wrote: Not currently. There is a bounty for the development of working Wildcard drivers for Free/Net/Open BSD. Care to write them? Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Moran Sent: Monday, March 29, 2004 6:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FreeBSD Do any of the Wildcards work with FreeBSD?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can i do voice chat without using the hardware
--- David J Carter [EMAIL PROTECTED] wrote: My aim is that, i want to connect my PC (where i installed the asterisk) to another PC in my network for voice chating. For this purpose, what are the steps to be done? which are the files to be modified. I would like to make use of the existing Hardware (sound card, network card etc), i am not using any extra hardware. Is X-Lite work in Linux? or any compatible s/w that works under linux? Have a look at these sites: - http://www.codepipe.com/id25.htm http://www.jaredsmith.net/misc/hgta/ http://www.wwworks-inc.com/asterisk/ http://www.fnords.org/~eric/asterisk/ http://bcwireless.net/moin.cgi/VoIPHowTo http://www.automated.it/guidetoasterisk.htm http://www.asterisk.org/index.php?menu=support http://www.voip-info.org/wiki-Asterisk+config+files http://www.voip-info.org/tiki-index.php?page=Asterisk If you have the CLI prompt then your almost there. If you have the audio set up in asterisk then you can use a headset/microphone to call the other party. CLIdial 1234 when finished CLIhangup Simple huh? Regards Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail - More reliable, more storage, less spam http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Shorewall and asterisk on Mandrake
I have your same setup: Asterisk running on a box that also runs SAhorwall. I can register to both WD and ICH. One thing I suggest is first getting Asterisk to work without shorewall. Next install the firewall but leave it wide open, close it down incrementally. Also turn on logging of every dropped/rejected packet and check the log file. If shorewall is getting in the way you will see the rejects to/from FWD or ICH in the log. Ask me off-line and I can send some config files but be warned they are more open then need be. --- Patrick Lidstone (Personal E-mail) [EMAIL PROTECTED] wrote: I am struggling getting asterisk to work on my firewall box. The Linux box is a firewall running Mandrake 9.2 and shorewall for security and NAT. Asterisk is compiled and running on the firewall box with a modified sample configuration. I am connecting to it using a Sipura on the local LAN. This works fine and I can phone between extensions (2201 and 2202) and access the voicemail menu via extension '8'. Now, I cannot get asterisk to register the two SIP providers I want to use: FWD and ICH. The log reports that it did not register - consequently I cant dial '6-612' to get the FWD date-speech. I've configured everything according to the manual and several example config files as referenced on voxilla. The error message I get is a timeout on sip-registration and some rtp timeouts. I assume its a shorewall issue. How do I need to configure Shorewall? (I have the following shorewall domains: net, masq, fw, loc used in the rules.conf) Does someone have a sample shorewall config? How can I easily tell that asterisk registered properly with the SIP provider? Could someone post some a current working sample configs for FWD and ICH which indicate the use of the various fields better than the existing samples: * For FWD I have 123456 (the number), AUTO_123456 (the user ID), password. * For ICH I have 1234567890 (the number without 1) 11234567890 (the number with 1), 98765432 (the user id), password. Voxilla doesn't mean anything to me, but I went through a similar learning curve a while back. The key to successful registrations behind nat (for me) are the following entries in sip.conf. My asterisk box sits on a natted network 192.168.0.x with address 192.168.0.5 ; ; SIP Configuration for Asterisk ; [general] port=5060 ; rtp port to bind to localnet=192.168.0.0 ; address space for local (natted) network localmask=255.255.255.0 ; netmask for local (natted) network externip=a.b.c.d ; a.b.c.d is public ip address of your router outside_addr=a.b.c.d ; as above bindaddr=192.168.0.5 ; where 192.168.0.5 is the IP address of your * box behind NAT nat=yes With these config changes, and asterisk restarted, you should be able to register ok (as reflected by sip show registry from command line. This is the crucial first step. In addition, for a bi-directional voice path you will typically require port forwarding of UDP traffic in the media port range specified in rtp.conf to the natted ip address of your asterisk box (192.168.0.5 in this example). A typical rtp.conf file might look like this: [general] rtpstart=50600 rtpend=50609 You should also configure your firewall to pass UDP traffic bi-directionally on port 5060. It is worth persevering - asterisk does work behind a natted firewall with the likes of FWD just fine. HTH Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Search - Find what youre looking for faster http://search.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Woodpeckers
--- Steve Underwood [EMAIL PROTECTED] wrote: A power spectrum plot will tell him he has a 60Hz hum. I think he already knows that. I think he can definitely consider solutions without following your suggestion. :-) No, It's not a 60Hz hum. Yes, 60Hz is getting into the line but the existing filters are removing the 60hz. What he hears is most likely 120Hz, 240Hz or something else or most likely a combination of various multiples for 60hz. I'd bet that the tiny speaker inside a telephone handset can not even reproduce a 60Hz tome. Yes you can hear a hum but it's the overtomes of 60 that you hear. Many people can not even hear down to 60Hz, some can but not everyone. If you were to design a filter wouldn't it be nice to know some thing about the noise? Is there a big peak at 360? how broad is that paek 5hz or 20hz? I would expect the power spectrum of a hum to have multiple peaks. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail SpamGuard - Read only the mail you want. http://antispam.yahoo.com/tools ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Woodpeckers
Get a spectrum analizer. Software will do it. Record the humming connetion to a file and then run it through software that plots a power spectrum. THere is plent of good open source software. Even some audio file ditors have this feature. You should be able to see the hum as periodic peaks at the multiles of 60hz. You can't even begin to talk about how to fix this without such a plot. --- Greg Hill [EMAIL PROTECTED] wrote: On Fri, 20 Feb 2004, matt wrote: I would have thought that should have been gotten rid of... Notch filter would only get rid of the fundamental frequency unless you notch at the harmonics also. Best bet I would say is a high pass filter at say 500hz. Reasonably simple to make with a resistor and capacitor... I think Sound On Sound magazine (which has archives on the web) had an article on how to build one... The thing is, most of the codecs available in * should be high and low pass filtering it anyway... My first thought was an RC filter, too. But I'd suggest that 500 Hz is too high a cutoff, because a note like a middle C is 256 Hz. I don't think it's uncommon for a voice (especially a male voice) to be in that range frequently. Although (in English, at least) vowels generally have a low frequency and sharp consonants (like a t) have a high frequency. It's the consonants which do the most for understanding a word, so maybe having those low frequencies attenuated wouldn't be so bad after all. But as somebody else pointed out, a first-order RC filter probably wouldn't attenuate quickly enough. By the time you get into a multi-order filter, it's probably best to be doing it in software (since the signal is going to be digitized anyway). Greg Matt Michael Welter wrote: I live at 8000' in the Rockies. We have lots of woodpeckers--they especially love to drill 4 holes in the north side of my house. They also like to drill on the arial telephone cables. Water then gets into the cable and causes a partial grounding on the circuits. This causes 60Hz hum to be heard on the line as well as a loss of amplitude. Qwest says tough s--t. All three of my POTS lines have hum. They are connected to an Adtran 750 and my asterisk system (a testbench for commercial endeavors.) The hum has always been bad on my end. Since I installed *, several of my callers have remarked about the hum. So here's the question: Could a notch filter of sorts be installed in the codecs I use? Filter-out everything between, say, 55 and 65Hz? Alternatively, is there a feature on the Adtran FXO card that deals with this? Thanks for your help, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail SpamGuard - Read only the mail you want. http://antispam.yahoo.com/tools ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone phones from FWD
It says good things about them that they've not charged you yet. I ordered a blue phone and had to wait. they kept offering me a black phone now but I said I'd wait as I didn't have time to test the phone and part of the test was to see if the blue was a good color. I finally accepted thier offer for a black phone and only then did my CC get charged. FWD was very respponive to e-mail. The order arrive two days after I sent mail saying OK send the black one I noticed they removed blue option from the order form. You might complain about the quality of the GS phone but not FWD's service. But they DO seem to have a problem getting these phones. It appears GS just can't build them fast enough to keep up with demand. --- Jason T. Nelson [EMAIL PROTECTED] wrote: I recently ordered a few phones from them for some testing I'm doing, but the charges still haven't appeared on my credit card nor have I received any confirmation email (not sure if I'll get one though). Does anyone know if they're severely backlogged for orders? -- Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/ BOFH Extraordiaire Sysadmin Ombudsman GPG key 0xFF676C9E GPG key fingerprint = 6272 5482 EDDD D0A3 FED2 262A FABB 599D FF67 6C9E disclaimer: My opinions are my own. Don't bother my employer about them. ATTACHMENT part 2 application/pgp-signature = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail SpamGuard - Read only the mail you want. http://antispam.yahoo.com/tools ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto apply a patch, diff file
You use the patch command. You can read the patch man page by typing man patch there are many options like for example to Save a copy of the original. It is also good to read the patch file before you apply it to get a feel for what will be changed. A patch file is simply a set of editor commands to be applied. --- Jan Larsen [EMAIL PROTECTED] wrote: Hi all I am new to linux and * , so could someone please explain how to apply a patch file (diff extension) Kind regards Jan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail SpamGuard - Read only the mail you want. http://antispam.yahoo.com/tools ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] festival voices
--- Tony Buser [EMAIL PROTECTED] wrote: Hi, I'm new to both asterisk and festival. I'm trying to figure out how to change the voice festival uses. For example, I've downloaded don_diphone to festival/lib/voices/english. I then edited /etc/asterisk/festival.conf and changed the festival command to: festivalcommand=(voice_don_diphone)(tts_textasterisk %s 'file)(quit)\n try adding a set of parens like this: festivalcommand=((voice_don_diphone)(tts_textasterisk %s'file)(quit))\n SNIP natural sounding voice? So far the best I've found were from here: http://hts.ics.nitech.ac.jp/download.html Have you seen festivox? It's a tool for building voices The key to making festival sound natural is to get the timming and entonation right. The astrisk app uses festivels demo test to speech application which is just that a quick dirty demo. Have you seen the markup language on the CMU site? http://www-2.cs.cmu.edu/~awb/festival_demos/sable.html Sable can do MUCH better then the simple tts application. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialing 800 numbers with VOIP
I think the please try your call again later part of the recording is to be taken seriously. I've tried 800 numbers and failed but I tried re-dial several times and it worked after a few attempts. I have no idea how my call is routed but I'd not be supprized if NuFone, Iconnect FWD and the like don't just have a small number of POTS analog lines for the outbound toll free calls and you simply have to be lucky to call when one of those is available. It may not be exactly that but I'll bet there is n of something and you are the n+1 caller. What you can do is put backups in the dial plan so if, say FWD fails you go for the next service in line. --- Tim Petlock [EMAIL PROTECTED] wrote: Hm. After seeing all the people who say it works, I thought - maybe I forgot to dial 9 in front of the number and that's why the call failed. So I looked up the Wells Fargo toll free number again and tried it. Failed. SIT tones and We're sorry, your call did not go through. Will you please try your call again later? The recording has nothing at the end that might give some clue who was generating it either. Okay, I thought, maybe it's a regional number and that's why. Nope, for further testing I looked up toll free numbers that for sure would be nationally dialable. They fail exactly the same way. Perhaps it's something in my Nufone account setup? I don't know. The SIP and IAX debug messages on my console don't look appreciably different from those where calls complete so I know the call is getting there. It's impossible for any recording to be generated on my * box because my sound driver has yet to work, it's not coming from there. I'm certainly not out to bash Nufone - I live in Uruguay and have a 64k internet connection. Vonage barely works because the codec they've chosen and the speed I've bought don't match up. (I imagine that it would work better if I had a 128k connection - but 64k is USD$42 per month and the providers prices increase exactly proportional to the speed you want.) There was about a 1.5 second delay in the other party hearing what I was saying and only one person could talk at a time. Using a different Cisco ATA to connect to * and then GSM over IAX to Nufone sounds night and day different. No delay, minimal distortion, two-way conversation - this is good stuff. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: Tuesday, February 10, 2004 12:50 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dialing 800 numbers with VOIP On Mon, 9 Feb 2004, Tim Petlock wrote: [ Long Explanation Deleted ] Nufone and Voicepulse would have to maintain some number of trunks with an ILEC or CLEC to complete toll-free calls. I dial 800 numbers all the time from my Nufone account without problem. Hell, my DID through Nufone -IS- an 800 number! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NIC card failure [was: System freeze]
VOIP is a very low data rate compared to the bandwidth of a switched 100BaseT network. Lets say you are using 100BaseT to trunk 100 simultainous calls at 64Kbps (How many of us really would ever do that?) 100 calls would be 6,400,000 bps. Well over what a T1 could handle but is only 6.4% of 100Mbps. or maybe 25% of the usable bandwidth of a switched 100BaseT segment. (Ethernet does not work well when loaded over 30% of it's nominal bandwidth.) Think of it another way 100BaseT is 100Mbps. If 100 calls are on the line each call gets 1000Kbps So VOIP in an office enviroment does not even come close to pushing the limits of Ethernet. Things like NFS and HTTP cascing with SQUID are much harder on a network and may push you to things like using gigabit ethernet but voice is a comparitively low data rate. How to select a NIC to use under Linux? Read the drivers. Pick the NIC with the best driver. Even a quick scan over the comments will let you see how much of the NIC's hardware is being put to use. Are the bytes being read one at a time with programmed IO? With DMA? Is there a hardware ring buffer being used? How about packet filtering on the card? Take a look, the Reltek is not a bad choise. --- Steven Critchfield [EMAIL PROTECTED] wrote: On Tue, 2004-02-10 at 07:01, mattf wrote: It is important to note that cheap nic cards that were really designed for 1% utilization on a workstation are not well suited for an Asterisk server installation with any kind of VOIP traffic. We foolishly put a Realtek card in a test server and after a month literally fried the NIC card, It was extremely hot when we took it out after seeing thousands of network errors. I would recommend a 3com 905CX NIC card, we have these in all of our Asterisk servers and they function beautifully, and I have yet to hear of one breaking down uder high traffic. Yes, they cost a lot more than your average $4 Realtek card, you can pick them up new for about $20-$30. Just to try and complete the message here, I have plenty of problems out of 3com cards. I was at an install where we had to jerk all the 3C905 cards and replace with Davicom cards. As for the Realtek cards, that is what is in my office switch with no problems for over a year. I wonder if the card was made by Realtek, or just the chip. I have had good experiences with the 8139 chip, and it may be just better quality cards. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialing 800 numbers with VOIP
I second the last comment below. I sent $5.00 via pay pal to NuFone and was up and runing very quickly. My problem is billing. What you do is send e-mail asking How much is left in the account? and then if it is low you send more. For test purposes this is not bad but I can't imagine setting up an end user with Asteisk and them explaining to them that for the rest of their life they will have to send account frequent ballance inquerries to NuFone. I could be wrong and maybe I'll get billed when my $5.00 runs out. I started with such a low amount just to see what happens when it runs out. Will there be an outage of service? Is that how I will know I'm at zero ballance? Maybe I'll get some e-mail or a voice message saying You're low, pay up. I guess I'll find out. But without a good billing system it can't be used in a small office that employs no phone geeks who will track ballances. About thier rates. Not bad. You need to ask and you'll get a spreadsheet with the per minute rates. They beat Iconnect by a slim margin. BUT they talk IAX2 which is a big plus. Why is this not on the web and automated? My guess is either a lack of devlopment resources at NuFone or he's more interested in supporting high volume customers who don't need a web site but more likely both. --- Robert Hajime Lanning [EMAIL PROTECTED] wrote: quote who=Joel Maslak I don't have a Nufone account (Jeremy - if you are reading - I would probably have one if there was a price for a starter package listed on your site - something for SoHo use, without any deep discounts or anything, just something to use to play with the service; I have a personal aversion to bothering with companies who don't list their prices), so I have no idea if Nufone's 1-800 service works or not. There are no fees. Think of it as a calling card. You put $10 into the account, then you have $10 of minutes. Per minute cost depends on where you are dialing to. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
Feb02 0:00 /usr/sbin/sshd root 325 0.0 0.9 1752 128 ?SFeb02 0:00 /usr/sbin/cron root 329 0.0 0.4 1488 56 tty1 SFeb02 0:00 /sbin/getty 38400 tty1 root 330 0.0 0.4 1488 56 tty2 SFeb02 0:00 /sbin/getty 38400 tty2 root 2609 0.0 0.2 2276 40 ?SFeb02 0:00 /bin/sh /usr/sbin/safe_asterisk root 2611 0.0 7.3 42144 1032 ?SFeb02 0:03 asterisk -vvvg -c root 2612 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2613 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2614 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2615 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2616 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2617 0.0 7.3 42144 1032 ?SFeb02 0:21 asterisk -vvvg -c root 2618 0.0 7.3 42144 1032 ?SFeb02 0:13 asterisk -vvvg -c root 2619 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2620 0.4 7.3 42144 1032 ?SFeb02 7:52 asterisk -vvvg -c root 2621 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2622 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2625 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2626 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2627 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2628 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c root 2629 0.0 7.3 42144 1032 ?SFeb02 0:00 asterisk -vvvg -c -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
I read a report of Asterisk running on a Microsoft X-Box. That's kind of a stunt as you could buy a decent PC for the price of a Linux-capable XBox. Id's still like to see Asterisk run on very low-end hardware The Snom IP phone runs Linux inside? I assume as Linux is GPL'd Snom will supply the source code? It would be fun to install an Asterisk server in a phone. --- Panny Malialis [EMAIL PROTECTED] wrote: Does anyone have it running on a Cyclades T100 ? same as used for ntop/nbox. I was thinking of using that as an IAX-sip translator for offices with NAT. CPU MPC855T (PowerPC Dual-CPU) Memory 32MB RAM / 4MB Flash (TS100) Interfaces1 Ethernet 10/100BT on RJ45 1 RS232 Console on RJ45 RS232 Serial Ports on RJ45 Looks like fun! Although a little lacking on memory. Any comments? Panny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Smallest server continued...
--- [EMAIL PROTECTED] wrote: This thread got me thinking of other servers that would run asterisk. The obvious question comes up if Xebian (the xbox version of Debian) would run as a SIP only server? Asterisk on an XBox would be a small box! Cheap too. It's been done. In fact by Mark hiom self if you beleive this URL that Goole found. http://216.239.53.104/search?q=cache:M1pPrvOlBewJ:nlug.org/mail/nlug__2003_12/0094.html+linux+asterisk+xboxhl=enie=UTF-8 But in my opinion Asterisk running on a Snom 100 would be even cooler and I can think of uses for it already. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip phones
The photo of the phone says Grandstream Budgtoe 100 when you click to see the larger image of the phone the text on the buttons becomes clear. They look to be selling aservice and the phone to go with it but I'd have to as my wife. She's fluent in Japaneese. I'm not even close. Hey Chinaman... I was wondering if the following SIP phone is just a Grandstream's OEM or just a japanese copy... http://sipphone.livedoor.com/ What do you think? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junk calls from FWD numbers
--- Rich Adamson [EMAIL PROTECTED] wrote: snip Actually, haven't had the problem here at all. I was about to write that I though I was getting these calls because my name started with A which put me at the top of the white pages listing and as I'm always on line I get to be the top of a lot of lists. But Adamson would be above Albertson unless Adamson is unlisted Now, I'm noticing that I have both Sue Albertson and Chris Albertson listed and it is Sue who gets 3x as many of these calls. Now I'm curious why some people _don't_ get these calls I'm in the FWD white pages. Are you people who don't get these calls unlisted? If so that would explain it. Does your system send incomming FWD calls to voice mail. If not them you'd only notice a junk call if you hear it ring. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What was the fix? (was good job on the list server)
Would you tell us something about the before and after setups? What was it that you did. Some of us who run mailmain list servers might want to know. --- Brian West [EMAIL PROTECTED] wrote: Rich, Thanks... I just double checked everything and she's still moving along. You shouldn't see those nasty virus stuffs on the list anymore either. :) L8tr, bkw = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk with big number of extentions.
Aside from the search time, which appearently grows linearly with the number of extensions, you have to worry about volitility in the list of 25K extensions. Lets say that every two years on average one of your uses changes. I think that means you will needs to make three changes a day. So you will be editing extension.conf and doing a reload a few times a day, every day. With 25,000 users will there _ever_ be a time when no one is using the system so you can do a reload? You will need some why to modify the dail plan without taking the system off-line.or you will greatly upset a lot of users. And while you are in there, it should not be hard to improve the search time better than O(M=N) at least for non-wildcarded extensions A simple binary search will get you to order of log2(n) which is a big deal with N 25K --- William Waites [EMAIL PROTECTED] wrote: On Thu, Jan 29, 2004 at 06:27:33AM -0600, Rich Adamson wrote: We are thinking of making network of about 25000 extension numbers. These extension will be SIP phones. Asterisk will be connected to some VoIP gateways through H323 which will allow to terminate calls. Can Asterisk handle such kind of load? No problem, as long as none of them make any calls. The number of extensions is relevant. I am not sure the level at which begins to matter, but the comment reproduced below from pbx.c gives some idea that eventually it may be an issue worth considering. It may be that at 25k extensions the O(N+M) search starts to become noticeable. /* * I M P O R T A N T : * * The speed of extension handling will likely be among the most important * aspects of this PBX. The switching scheme as it exists right now isn't * terribly bad (it's O(N+M), where N is the # of extensions and M is the avg # * of priorities, but a constant search time here would be great ;-) * */ But of course there are ways around using ENUM and the like. -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Introducing Firefly
When will the IP phone be available and would you have an idea about the price? --- Samuel Jimenez [EMAIL PROTECTED] wrote: Nice!! Have just tried it a bit, seems cool... Congrats!!! Will test it against my * box and will provide some feedback. Thanks! Sam\\\ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk with big number of extentions.
What are you talking about? You can already reload the dialplan without affecting existing calls. Thanks, Now that I know it can be done, I'm sure I can figure out how. I just figured that a re-load zapping a media stream going through Asterisk is just normal and to be expected. Din't even think about it being either a bug or operator error. But for my use this is not important as at most I'd be running a small office with ~10 lines were people go home at night. But with 25,000 active users ...when convenient would be a long, long time. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream 100 sidetone
Stephen I think hit it on the mark. I could not figure out how sidetone could be heard as an echo and how it could be loud. On my GS phone the sidetome is very low but with zero delay. The best way to test the side tome is to talk to the voicemail or record application running on a local Asterisk server. Record does not send audio back so anything you e hear is real sidetone. I find it to be decent quality but to low in volume. --- Stephen R. Besch [EMAIL PROTECTED] wrote: dkwok wrote: For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the volume of the voice call. As the sidetone is louder than the conversation it is getting rather distracting. Can the sidetone be calibrated or adjusted? If not, how are people coupling with it? If I'm not mistaken, what you are calling sidetone (the copy of your owm voice that is played back to your earpiece - it's reassuring to hear yourself talk) is actually real echo generated somewhere other than in the phone. It is a network issue, not a phone issue. Read the many, many post on echo and visit the WIKI. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapateller
I have a few (more specific) questions about 'Zapateller': 1) How would you test this?? Would I need a predictive dialer machine like the telemarketers use. OK, I could just wait and see if it seems to cut down the unwanted calls but that's not really a test. 2) I don't understand how I would choose to use the answer option or not. under what conditions would answer by better then not using the option? I asume if I used answer the tones would play in caller's ear and be _very_ anoying to them. Why would I want to do that? There must be some reason are why would the option exist? --- Steve Foy [EMAIL PROTECTED] wrote: Hi, I'm just wondering about 'Zapateller'. How exactly does it work!? I might be interested in employing it at work here, but wondering if anyone's using it? echo*CLI show application Zapateller -= Info about application 'Zapateller' =- [Synopsis]: Block telemarketers with SIT [Description]: Zapateller(options): Generates special information tone to block telemarketers from calling you. Returns 0 normally or -1 on hangup. Options is a pipe-delimited list of options. The following options are available: 'answer' causes the line to be answered before playing the tone, 'nocallerid' causes Zapateller to only play the tone if there is no callerid information available. Options should be separated by | characters -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Junk calls from FWD numbers
My Asterisk server registers two FWD numbers. On average I get about one call a day from someone calling from an FWD number and leaving a pointless, under 10 second message. It's easy to see who these people are if I look in my CDR file I can see thier name and number. They seem to be new FWD users, likely who've just downloaded FWD's Xten softphone and then dial some random FWD user (me) to try it out. I wonder if these same people when they first got a POTS phone installed in thier home got out the white pages and dialed randomly asking anyone who'd answer Hi does this work? can you hear me? Question: Does everyone with an FWD number get these junk calls or am I the only lucky one? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has Nufone gone belly-up
I think discussions about which VOIP service providers are best or not best is a reasonable topic for an asterisk user's list. After all, selecting a provider is a big deal Rants about some company just make the rant-er look bad and should be avoided. Back to NuFone. I've just started testing thier service. So far so good. The sound quality and pricing is very much like Iconnect but NuFone will accept IAX2, a big plus. But my e-mails are ignored. Actually sending e-mails was part of the test of this vendor. I'm testing to see if they answer e-mail. So far, not good. Maybe NuPhone needs to loosen up their spam filters or at least scan the spam folder every few days. --- Brian West [EMAIL PROTECTED] wrote: Have you tried to call them? Your emails could have been caught up in a spam filer or such I use nufone daily for our 888 service. I talk to Jermey daily. So I dont know what your beef is but your rant has no place on this mailing list if you are having problems and have spent any time trying to get someone on the phone you must be doing something wrong. I do know about 2 weeks ago GoDaddy screwed up nufone.net's domain and it was sent off into LALA land for a few days. So please keep your rants off the list. bkw On Sat, 24 Jan 2004, Sathya wrote: Folks, I've ordered a new account from Nufone last month. Transferred money to Nufone through their paypal account. I had communication with Nufone sales up until two weeks back. Since then there were no replies to my emails. I am afraid with this kind of unresponsiveness how one would run a reliable service with this company. Have no bad feeling with Jeremy as the author of widely used h323 channel, but my concern is about the company NuFone. Lot of newcomers when asked for IAX termination/Origination we say NuFone. I just want to record my experience so far, as it would help anyone wanted to start with this company. I can live with the fact that they do not have any web based interface for customers to do anything with the service as claimed by the website. But cannot understand taking two weeks to answer a freaking email. (Well in the absence of trouble ticketing system or web based access to accounts, email is the only way to contact Nufone) I have services running with Iconnect and Voicepulse etc and I was just trying to use Nufone being well recommended in this list. I am not here to tarnish Nufone name but I have no option but to ask the community since there is no response to my emails or there is no indication of when my service is available. If they have gone belly-up, well I can then concentrate on some other company and consider my money as a cost of a bad choice on my part. If I am a very rare case who just had a bad experience with an excellent company ( I wish ), Nufone please fix this ASSAP. Later Sathya ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Subject: Re: [Asterisk-Users] Grandstream 100 sidetone
--- dkwok [EMAIL PROTECTED] wrote: Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 I've got the same firmware version. So it appears that sidetone volume is not dependent on the firmware version. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 ATTACHMENT part 2 application/x-pkcs7-signature name=smime.p7s = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Subject: Re: [Asterisk-Users] Grandstream 100 sidetone
--- dkwok [EMAIL PROTECTED] wrote: Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 I've got the same firmware version. So it appears that sidetone volume is not dependent on the firmware version. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 ATTACHMENT part 2 application/x-pkcs7-signature name=smime.p7s = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream 100 sidetone
What firmware version do you have? On my phone the sidetone is very weak, I have to listen carfuly to hear it but there is no echo and the sound quality is good. It's to bad GS doesn't open source the software inside. I'd fix it in a minute by putting a volume setting on the admin web page. --- dkwok [EMAIL PROTECTED] wrote: For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the volume of the voice call. As the sidetone is louder than the conversation it is getting rather distracting. Can the sidetone be calibrated or adjusted? If not, how are people coupling with it? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 ATTACHMENT part 2 application/x-pkcs7-signature name=smime.p7s = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream 101
--- dkwok [EMAIL PROTECTED] wrote: Just got GS 101 phone and plugged into the network. Peoplehere complain about these phones but I don't seem to have a problem, well not after getting them set up correctly. I'm running with Software Version: Program--1.0.4.39Bootloader--1.0.0.13 HTML--1.0.0.20 Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. I'm using SIP info also with payload type set to 101 2. there is no sound coming from the other end. For some reason I found I had to place the disallow=...allow=... stuff under [gs] putting it in [General] didn't seem to do the trick. I also put reinvite=no in [gs] I once had sound going only one way due to t stupid error in my firewall config. I was purposfully droping packets and logging each one of them. Are you running firewall software on your * server? ethereal or other ethernet sniffing software is usfull to debug this kind of stuff I have a sip.conf setup for GS: [General] disallow=all allow=ulaw allow=alaw [gs] canreinvite=no dtmfmode=info In the GS101 setting rtp port = 5004 sip port = 5060 dtmf = sip info codec = pcmu codec = pcma Any pointer of a sample of config file would be most appreciate. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 ATTACHMENT part 2 application/x-pkcs7-signature name=smime.p7s = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Standalone FXO device
I connect with the Clipcomm device, my DSL gets down and it gets up only when I switch remove the line from the device. I dunno what kind of problem it is. With DSL one line shares voice and data. _Every_ device plugged into your phone line except your DSL modem needs an in-line filter. The filter prevents the non-modem from either changing the line impedance or putting high frequency noise on the line, either of which will kill the DSL signal. So if you plug an FXO device into the same line that also carries DSL and don't use a filter the FXO device could very well kill your DSL connection. But it all depends, some times you can skip the filters. I did but then I plugged in one more analog phone and broke DSL. I installed a few in-line filters and now the DSL works better. If you did place a filter between the clipcom device and the rj11 wall jack then something else is going on = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mailing List Lag
Maybe the words nine million was not ment to be taken literally. What if he said about a gazillion Then we'd all be arguing if gazillion == 1x10^14 or 1x10^16 Have you ever set up mailman on a Linux system? 9,000,000 would be a real trick setup not something you'd do with a standard PC. You have a rack full of mail servers, raid disks and the works. Lets do some math: There are 60*60*24 seconds in a day 9E6 per messages/day means a little over 100 messages per second sustained 24x7. I really doubt it. I think nine milion is a technicl term just like sh*t load or way lots. --- Philipp von Klitzing [EMAIL PROTECTED] wrote: Hi! 900 mails / 150 per day = 6 Subscribers.. Is the Asterisk community anywhere near that big?? There is more than just one list on that MLM, and you'll probably have to count in adminstrative messages (?) and maybe even error messages. Still that would reveal a huge number of subscribers... Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium X100P for $43
Pirated?? I don't think Digium makes the X100P cards. They are common cards sold under several brand names Digium being just one of the names. The selling price varies dramatically depending on the re-seller and which brand name is on the outside of the box. What Digium does is bundle in one hour of consulting with each card, that's why the high price. You are paying mostly for theconsulting time. You would have to be nuts to use a pirated card. For saving a trivial amount of money? Stupid idea. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Digium X100P for $43
You _can_ copyright the artwork on a card. (The artwork defines path of the printed conductive traces.) If you own the copyright, well, you own it. But does Digium own the copyright? In the case of the X100P they don't. The X100P is re-branded comodity hardware. Of cource they all have the same FCCID number. If they are zapata designs then the design is in the public domain and _anyone_ may do as they please with it. See the quote ...and are perpetually placed in the public domain... at this site http://www.zapatatelephony.org/ --- Sean Cheesman [EMAIL PROTECTED] wrote: If they're truly counterfeit, they need to act! One call to eBay will solve it. They're even taking pre-orders on the T400P and E400P that they're getting ready to build. $1050 a pop. Digium should get involved if this is truly a problem and not just bad ethics. Sean -Original Message- From: Dustin Goodwin [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 11:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Digium X100P for $43 It's funny this is the second hardware counterfeiting story I have heard this week. What is going on? - Dustin - Sean Cheesman wrote: for the record, mine has the same fcc id number as the Digiums. Is this typical for copied hardware, or is there something a little fishy going on here? -Original Message- From: Doug Meredith [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 7:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Digium X100P for $43 SamW [EMAIL PROTECTED] wrote: Digium X100P / new cards are is available on ebay for $43. Actually they seem to be made by Digit Networks. Doug ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium X100P for $43
One interesting quote from the zapata web site: These cards are currently available from Linux Support Services in Alabama. They are now in full production, and the cards are readily available. For information on pricing and availability, please contact Mark Spencer ([EMAIL PROTECTED]). For those wishing to fabricate their own cards, the gerber photoplot files may be downloaded here. The designers of this hardware seem to have worked in the spirit of the open source movement and purposefully made thier CAD files public. I think it's great that we have an Open Source software system that can run on open source hardware. Actually while it says public domain on the web site. The PDF files containing the design say GPL. --- Adam Hart [EMAIL PROTECTED] wrote: They ain't counterfeit, Digium thought best not to reinvent the wheel, this is the price of that decision. Can we take the red (or what is blue) pill, forget this ever happened and continue supporting Digium. - Original Message - From: Sean Cheesman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 4:44 PM Subject: RE: [Asterisk-Users] Re: Digium X100P for $43 If they're truly counterfeit, they need to act! One call to eBay will solve it. They're even taking pre-orders on the T400P and E400P that they're getting ready to build. $1050 a pop. Digium should get involved if this is truly a problem and not just bad ethics. Sean -Original Message- From: Dustin Goodwin [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 11:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Digium X100P for $43 It's funny this is the second hardware counterfeiting story I have heard this week. What is going on? - Dustin - Sean Cheesman wrote: for the record, mine has the same fcc id number as the Digiums. Is this typical for copied hardware, or is there something a little fishy going on here? -Original Message- From: Doug Meredith [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 7:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Digium X100P for $43 SamW [EMAIL PROTECTED] wrote: Digium X100P / new cards are is available on ebay for $43. Actually they seem to be made by Digit Networks. Doug ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ADSI phone vs. IP phone
One other option, although I hate to suggest it but will admit to having done it a few times... Ethernet uses only two twisted pair. Cat-5 wire has four pair inside. You can crimp two RJ45 terminals on one lenght of CAT5 wire. It works and remains CAT5 if you keep the pairs twisted all the way into the terminals. OK, yes it's unclean and ugly but the uglyness is inside the wall in back of a cover plate and the user sees two RJ45s where there was one before. That said, last time I got to wire a new office we pulled double the number of Cat5 as required but only terminated it as required. in either rj45 or 11. You might concider using wireless for the phones. Ether a wireless LAN or just wireless analog phones. Ether way no wires. As I remember the average cost to pull a wire ad terminate both ends is about $100. cheaper than a wirelass adaptor. --- [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dustin Goodwin Sent: Monday, January 19, 2004 11:18 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ADSI phone vs. IP phone Why wouldn't you just use your existing Ethernet infrastructure putting the IP phones inline between the wall jack and the PC? There are a number of IP phones that have builtin switch/hub that allows the PC to daisy chain off the IP phone. Probably because it's well known that these setups are prone to failure of either the PC's connection, the phone's connection, or degredation of one/both. It also breaks switch envirenments where spanning-tree portfast is enabled (not as big of a deal if the deployment is in concert with the infrastructure group, as it should be). = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk
Yes, you can keep non-authorized SIP callers from accessing the PSTN by setting up the .conf file correctly as below but you can also run a fire wall on the box that Asterisk runs on. Firewall off SIP ports except for if they come from your SER server. This will work even if Asterisk is broken or misconfigured. Security sould always be applied in multiple layers: use both a belt and suspenders I like the shorewall firewall script. configuration is conceptually easy it uses the cisco-like idea of zones. --- Fran Boon [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: I'm trying to bundle the powers of Asterisk and SER. Asterisk for pabx functionalities and termination to landline/PSTN, and SER as SIP Gateway/Proxy. With my current configuration the SIP user just adds 0 as a prefix to a number, and the call will go out to PSTN over Asterisk. For this to work I added the rewritehostport() function in SER to point to the Asterisk IP (different from the SER ip). At the moment I just added the following line to my sip.conf (in the [general] section): context=from-sip But my question here is, everyone can (ab)use this by connecting directly to the Asterisk IP. This way they can easily dial out over the PSTN network. Hi, This sounds a very similar problem to me, despite the different context. The 'default' context in the [general] section shouldn't be (ab)usable - set this to something like [bogon-calls]. Then set up a specific peer lower down: [ser] context=sip-legal host=y.y.y.y ; IP address of SER Se this Wiki page for more flesh of my (not yet fully working!) configs: http://voip-info.org/wiki-Asterisk+cisco+FXO Good luck! Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Hardware for Asterisk
--- Cees de Groot [EMAIL PROTECTED] wrote: Steven Critchfield [EMAIL PROTECTED] said: In general, you get what you pay for, and less so when you go bargain hunting. It all comes down to the same old problem of figuring out what your time and downtime are worth. Steven, I think this is one case where the bargain hunter wins If that Dell server really is $318 delivered and comes with a 2.4ghz Pentium and 128mb ECC RAM and on-site maintanance I'd say buy two and keep a hot preconfigured spare. Oh and they can come with RH9 Linux pre-installed. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware for Asterisk
Software RAID vs. Hardward RAID??? There IS no Hardward RAID it's all software the difference is only where the software lives, in ROM on the controler card in the RAID box or in a Linux driver. If you go top of the line and buy a Netapp network attached storage box. It thin it is just BSD running on Intel hardware but all closed up so it looks like a turn key system. Same with Sun. Thier hardware RAID has a SPARC CPU in the raid box. For Asterisk all you would need is a simple disk mirror at most. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ultra-cheap asterisk box
I'm looking to do about the same thing, build very low cost systems. (I'm looking at putting Asterisk at some non-profit organizations.) but one thing you can't make a compromise on is reliabilty. It has to work and keep working for years to come. I was able to keep the price of a new PC to about $300 ad still use an ASUS mainboard and an AMD XP2600+ The trick is to add absolutly nothing not needed. No floppy, no CDROM so you can run off a 200W P/S. Next I'll experiment with a notebook sized IDE disk drives and to see if _underclocking_ the CPU reduces it's power comsumption enough that we can save one fan. Ideally Asterisk will be ported one day to Linux/ARM or some other very low cost platform. for VOIP you do not need the PCI slots. In theory Asterisk could run on a Lynksys router box with re-flashed EEPROM. After all Lynksys' latest wireless router runs Linux inside Low cost to me means low total cost of ownership To get this I don't think buying the lowest priced parts is the way to go. I want quality mainboard, and a quality power supply and, this is importernt: A low internal case temperature. for this reason I'll spend the extra $50 to go with Antec cases and ASUS mainboards over the generic ones. What I'm finding is that the PCs are so cheap that the cost of electric power to run them is now a large part of the cost. (assume 0.20/kwh times 200W times 365 days = $350. So you pay for the PC again every year in electric power to run it. Worse. In an office with airconditioning _all_ of that PC's 200W goes to heat and your A/C unit will use about 220W of power to remove that 200W of heat.) and at a small office they will not have a server room so noise from the fan is an issue. --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 (¤ 216) plus a small ¤50 drive and cables, so say ¤300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP phones and potentially a fax solution? any ideas? thanks roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re hardware requirement - asterisk
--- [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have just checked the Openbsd box on the if interface. fxp0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:01:02:78:11:e8 media: Ethernet autoselect (10baseT) status: active inet 203.219.167.126 netmask 0xfffc broadcast 203.219.167.127 inet6 fe80::201:2ff:fe78:11e8%xl0 prefixlen 64 scopeid 0x2 For fxp0, the internal interface although the nic can do full-duplex it seems to me that it is only running simplex!! Why do you think it is running simplex. I read the above and see where it says (100baseTX full-duplex) I don't think 10BaseT can run full duplex. I could be wrong but I don't think so. But why does it matter? A single VOIP connection will not even use 1% of a simplex 10BaseT. Simplex 100BaseT should be able to handle dozens and dozens of calls Same for xl0, the external interface. It is running 10BaseT but again it is simplex. Does that affect my voip performance? Is it true that every step of the way the network has to be full-duplex? David Kwok = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capacity testing
/tls/libpthread.so.0 - If anyone has tried something like this or has any comments, I'd be interested in hearing from them. jesse ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware for Asterisk
--- calvis [EMAIL PROTECTED] wrote: I am real close to finalizing my hardware selection for my Asterisk test machine. I am going to use the following hardware: Dell 400SC w\Red Hat 9.0 1 - 4 Port TDM40B Card (FXS) 3 - Wildcard X100P Cards (FXO) It does not matter if the PC is a Del, Compaq or you built it yourself. What matters is the mother board that Del is using. Find out if there is a way to assign each Digum card it's own interrupt. Don't bother with the RAID controller it will not work with Linux. but Linux has it's own RAID in software. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] capacity testing
I am rather curious as to why I seem to be using up all memory although I am not running any unnecessary processes, or should I actually disable all modules, other than really necessary ones to support VOIP? Do you mean that Asterisk is using up all of your memory or that all of your memory is i use? If the former, that's odd. But you should expect the later. Linux is designed such that it will always try to put almost all of your memory to good use. It will use extra RAM as a disk cache. So if you were to cut the amount of RAM in your system in half you'd have only slightly less free RAM as Linux would use a smaller disk cache Now if you say the Asterisk process in the idle state is 450MB then something is wrong = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: hardware requirement -asterisk
--- [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Referring to my previous post about degradation of voice quality when having more than 2 connection. The actual route is: pc xlite - local asterisk box - iaxtel - local asterisk I have tried out a different situation: pc xlite - local asterisk box - iaxtel and the second connection pc xlite - local asterisk box - iaxtel - local asterisk The same degradation happens as soon as the second connection is connected. I am suspecting the ADSL connection. The internet part is ADSL with 512k down and 128k UP. The nic is a 3c905c 100baseTX and connected to a NEC ADSL modem. # ifconfig xl0 xl0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:01:02:78:11:e8 media: Ethernet autoselect (10baseT) status: active inet 203.219.167.126 netmask 0xfffc broadcast 203.219.167.127 inet6 fe80::201:2ff:fe78:11e8%xl0 prefixlen 64 scopeid 0x2 But ifconfig seems to suggest that it is running in simplex mode. Is the degradation a result of the ADSL connection? The fact that you 10BaseT is simplex will not matter as 10Mbps is 20X faster than even the ADSL's downlink speed of 512kbps You bottle neck is the 128kbps uplink speed. You'd think that is a lot but you can't got say codec X uses Y bits per seciond, so two calls are 2Y bps and keep adding calls untill your 128bps is full. It don't work that way. Think in terms of _probibilities_. Say your uplink is one quarter full. What does that mean? It means it is running at 128kbps 25% of the time and zero 75% of the time. So if an audio packet is placed on that line there is a 25% chance it will be delayed in an outbound queue. It is those delays that you hear. Actually the amount of delay is a distribution and what you hear are the tails of the curve. (i.e. there is a 25% change of a delay then there is a 12% change that two packets back to back will be delayed, 6% of three and so on.) With one audio stream there is no competition for the uplink. Adjusting the packet size can have an effect. Very long packets are not good = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hardware requirements of asterisk
He sais sound cards so I assume he is using softphones or possably the console phone. But then he sais two users making call Does this mean four phones? if so and you are using sound cards how is this done? Details please. --- Rich Adamson [EMAIL PROTECTED] wrote: I have been playing with 2 Asterisk boxes for testing purposes, it has been going very well. The 2 boxes are PII celeron 400 (HP Deskpro) with sound cards and lan. I have iax connecting the 2 boxes. For making cals and testing out recorded message for 1 connection it was working quite well. However, when I stressed it a bit with 2 users making calls, we started to here voice degradation and cracking noises. However, top shows cpu is 94% idle. I am suspecting the network. However it is 100M switch and I have not had any clue. I suppose it should at least be able to handle 10 calls similtaneously for even a small office. So what is the recommended spec for 5 users or 10 users? Without any other factual detail, best guess is half vs full duplex problem on one or more of the devices (phones, PC, etc). Assuming you're using sip phones for testing (and we really don't even know that for sure) and depending upon exactly what parameters you've applied for each sip phone definition within asterisk, calls between phones are set up by asterisk. Asterisk then instructs the two phones to communicate between themselves, and bows out of the audio session. So, if you really are using two sip phones, then you have a networking problem between those two devices and not with asterisk. For anyone to offer suggestions, you really need to provide more facts. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: failover (was Re: voicepulse)
I'm having the same concerns. What we REALLY need is the ability to test the exact nature of the problem. OK We could use SER to front end SIP calls but Asterisk should report the problem and allow the dial plan to test it. It's a needed missing feature in * What about AGI? I don't know much about AGI yet but it may help solve this problem. --- Matt Lawson [EMAIL PROTECTED] wrote: But this is not to say _you_ can't built a reliable VOIP based system. Get _two_ providers and set up your dial plan in extensions.conf to fail over if one service fails to connect to dial via the next one and finally if both fail use pstn. your users will see a system the just works. Now there's an idea. I'm playing with this now, but there's at least 1 case I'm having trouble recognizing: The call connects but then drops due to unauthorized. It then only goes to the h extension and I don't get a chance to try again. Is there anyway to detect this? I have to cover all of the following cases: 1. VOIP IP address is not reachable. Goes to extension n+101 (seems to work as expected) 2. VOIP service answers but refuses with call with unauthorized. It just goes to the h extension Is there any watch to catch this failure? Perhaps put a timer on it and say if the call was less than 5 seconds or something try the next one? Yes I am using a correct username and password and getting this today (not from Voicepulse, from another provider). But there's also a moderate chance that during our systems' setup a name or password could be misspelled so I need to cover this case. If your providers requires a pre-paid account the the account bvallance runs out then I gues you'd get unauthorized. So this could be a real case that will happen 3. VOIP service connects but reports all busy. Well this one is hard to test. But I can make the Zap channel busy. It goes to extension n+101 as expected, so I'll have to assume that a busy VOIP service does the same thing. I get this from the stwo VOIP providers I use about 20% of the time. I guess they have only so much gateway hardware. Normally a quick re-dail does it. I was trying to determine if the t or h extension would be useful for these but I think not. The timeout has to be set long enough for someone to actually answer (20-60 sec or whatever). The h is always visited at the end of the call, whether it was sucessful or not. Any other cases, or suggestions how to handle case #2? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do we updated to the new .7.1 version.
Step #1. Go to your local landfill or dumpster and aquire another PC. Any old junker will do. Install Linux on it. This will be your test machine. (Seriously, you'll want at least a 200Mhz Pentium class box with about 128MB RAM and a 5GB drive.) Use the test machine to try out stuff. A mistake on the test box will cause no harm. Never do development or test on a production server. Step #2 Assuming you've set up thre test system. Build the new 7.0 code just like you did the 5.0 code. the make install will overwrite the old stuff with the new stuff. If you have any problems post a _specific_ question to this list. --- Ariel Batista [EMAIL PROTECTED] wrote: Yes folks it's me a Newbie. Remember I am also a non-Linux person trying to learn. I have a production Server running Asterisk .5 12/02/03 CVS, and would like to upgrade it to the new .71. Has anyone come up with instructions (Documentation for us newbie) on how to do this? My server is running Mandrake 9.0 which I know nothing about! Sorry if this sounds stupid but all the instructions I found were for CVS's. And since it's a system that has been running fine I have not wanted to do any updates. But I figured that a new release is better then the .5 I have. (I have some small Zap port problems).SNIP = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone.net net wackiness?
Looks like they went off the air just after my PayPal payment was processed. I gues we wait a couple days to see if Nufone has gone belly up/bankrupt/gone or if this is just a domain name screw up. --- Steven Critchfield [EMAIL PROTECTED] wrote: On Tue, 2004-01-13 at 01:26, Brian Capouch wrote: I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a parking site. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux Distribution
I have to agree with the below but only if it is an answer to the limited question of Which is best to use for my Astrisk server. For a server you are using such a small percentage of the Linux distribution that they are effectivly all the same. A server will not make us of any of the graphical interface or Desk top software. Most * servers run with no keyboard or CRT plugged in. BUT, If you are running an Asterisk server you will likley also have a Linux box for development, testing and general e-mail and web serfing. For this purpose it does matter, a little. They all will do the job but differ in terms of the details of exactly what software is included and how the menu system on the desk top is set up. Still none is better but they are differntent enough that people can have strong prefference. The differences between distributions are minor. I doubt an inexperianced user could tell this Solaris 9 box I'm writing this on from a Linux system. Both run gnome and look the same on the surface. But Linux and Solaris are far more different then any two Linuxes. That said, pick a desktop system you like. You can get a free download of any of them or low priced CDs at cheapbytes.com and try them out. Then use the same distribution for your server. They're all just Linux. There is no best. This question is asked so frequently it almost looks like a troll to me. :) I've therefore updated the FAQ on the wiki: - http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ Which Linux distribution should I choose for Asterisk? -- There is no best distribution. There are no fundamental differences in functionality or behaviour between Linux distributions like there are between versions of Windows. Pick whichever one you feel most comfortable with. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.0
I think the exchange below shows us that before 0.8.0 comes
out, maybe there should be a 0.8.0-beta then after no problems are
reported in a few week period a 0.8.0-release candidate and
ten 0.8.0 itself.
It's hard to call a realease stable until a number of people outside
the developer's lab have used it for a while.
The other idea is that everyone just knows that x.y.0 == beta
and they all wait for a .1 or .2 realease.
--- WipeOut [EMAIL PROTECTED] wrote:
Tilghman Lesher wrote:
On Tuesday 13 January 2004 00:10, Mark Spencer wrote:
Okay, it's 15 minutes late, but it's out, thanks very much to all
the
people who worked so hard this weekend to make this possible!
There is one bug so far and it's critical. It breaks includes and
the
GotoIfTime application. I'll own up to writing the broken code.
The
fix is very simple, though (attached).
-Tilghman
Index: pbx.c
===
RCS file: /usr/cvsroot/asterisk/pbx.c,v
retrieving revision 1.92
diff -u -r1.92 pbx.c
--- pbx.c11 Jan 2004 09:19:16 - 1.92
+++ pbx.c13 Jan 2004 07:21:12 -
@@ -2922,7 +2922,7 @@
return;
}
-#if 0
+#if 1
s1 = s1 * 30 + s2/2;
if ((s1 0) || (s1 = 24*30)) {
ast_log(LOG_WARNING, %s isn't a valid star time. Assuming no
time.\n, times);
Why not quickly patch the source an release 0.7.1 if the bug is
critical?
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Re: [Asterisk-Users] Asterisk and Festival (* dies with no info)
dump, I never see one generated. Any thoughts on what might be happening here? What am I doing wrong? -- Doug ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How does the PSTN termination indurty realy work?
Does anyone here understand how these VOIP/PSTN gateway companies work? There seem to be a number of small outfits that offer about the same service. I'm thinking that a company that is really just one person and a web site can't possibly own and operate gateway equipent in 100 contries world wide. I figure there must be some large outfit who _does_ have physical equipment installed in hundreds of locations that leases time to re-sellers like Nufone, Iconect, Voice Pulse, Addaline, xvoip and the others If this is realy how the industry works, then does it matter which retailer you pick? Matter in terms of audio quality and reliabilty, this is. I know each will have it's own customer support and pricing. This on the heels of switch-1.nufone.net being missing out of DNS. We have customers that expect their VOIP to work. Is there anybody that's reliable? I am having probelms connecting to voicepulse this morning. Is anybody else = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicepulse
--- Steve Sobol [EMAIL PROTECTED] wrote: Matt Lawson wrote: I was just about to write the same thing. It says busy. Is is REALLY busy or is something else wrong? This on the heels of switch-1.nufone.net being missing out of DNS. We have customers that expect their VOIP to work. Is there anybody that's reliable? I've been doing some testing and so far I'm not 100% impressed by the VOIP services I've seen. They provide a good service but my local phone company and ATT longdistance service is more reliable. But this is not to say _you_ can't built a reliable VOIP based system. Get _two_ providers and set up your dial plan in extensions.conf to fail over if one service fails to connect to dial via the next one and finally if both fail use pstn. your users will see a system the just works. About Nufone's problem. I bet they'll start thinking about getting a backup DNS service and maybe geographic deversity. A company should be able to even stay on the air if there is a server room fire using techniques like round robin DNS and West cost and East coast servers run by different, unrelated hosting companies. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VOIP-PSTN service recomendation?
I'm looking for a service that will accept VOIP calls and send them to the PSTN. Or, I should say _another_ service that will do this. I don't need the other direction Currently I'm using IconnectHere and it works, but I get complaints of poor audio quality from the other end. But it sounds OK on my side. I like Iconnect's price model: very low/no monthly fee with a samll per minute charge. I'm in US 310 area code and call to US, Japan and Canada. Who are Iconnect's competitors? Anyone want to recommend a service? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */
Anyone who does not like the fact that all code must be disclaimed, sent through Digum to CVS and that GPL'd code can't go in can fix that problem. All you need to do is copy the current CVS and use that to start your own project. You can call it Asterisk Prime or Star and make up your own rules within the limits of the GPL. So there, if you want the rules changed _you_ can change them. But I'll bet you don't want the rules changed so badly that you would go to that much trouble. I and I assume most everyone here whould rather put up the way it is than duplicate Mark's efforts --- Steven Critchfield [EMAIL PROTECTED] wrote: On Sat, 2004-01-10 at 18:47, [EMAIL PROTECTED] wrote: I have always been suspicious of centralized control and dictatorship, benevolent or otherwise. After thinking for some time about the licensing structure of code for Asterisk, I am not sure that their motives are so innocuous and altrusitic, or at least this is not reflected so well in the fine print. After learning that all code must pass through Mark, I am even less sure. It means that Digium remains in a position of control and dominance over what is ostensibly communal property. I seem to remember at one point that all code in the official linux kernel had to go though Linus. Did we suffer? I don't think so. All code going through Mark isn't a bad thing. If you look through the cvs logs, you might see there are 3 or so commiters right now. I know jeremy is able to commit, but I think he is limited(probably self imposed) to theSNIP = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
--- Steve Underwood [EMAIL PROTECTED] wrote: WipeOut wrote: Granted five 9's is never easy but in a cluster of 10+ servers the system should survive just about anything short of an act of God.. You do realise that is a real dumb statement, don't you? :-) A cluster of 10 machines, each on a different site. Guarantees from the power company - checked personally to see that aren't cheating - that you have genuinely independant feeds to these sites. Large UPSs, with diesel generator backups. Multiple diverse telecoms links between the If he says cluster he likely means 10 servers in one rack. But still you are right. It is all the other stuff that could break. You will need paralleld Ethernet switches (Yes they make these, no, they are NOT cheap.) you will need some kind of fail over. The switches can do that for you. (do a google on level 3 switch) It's the level three switches that make .9 possible but half or more of your hardware will be just hot spares so it really will take a rack full of boxes Each box should have mirrored drives and dual power supplies and each AC power cord needs to go to it's own UPS Has anyone tried to build Asterisk on SPARC/Solaris? One SPARC server is almost five nines all by itself as it can do thinks like boot around failed CPU, RAM or disks. I've actually pulled a disk drive out of a running Sun SPARC and applications continoued to run. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IConnect audio quality
Hello, I've subscribbed to IConnect. I use it eclusively for outbound calling. I like the rates they charge but people I call complain about the audio quality. They say it sounds like I'm using a cheap mic. or they complain about echo. The sound is very clean at my end. I'm using a Bundgtone phone with meadi routed through Asterisk to IConnect. It's not the BT100 phone as the Audio is OK in cases where I don't use Iconnect. Question: What service (other than Iconnecthere) should I consider moving to? Here is what I want: * Acceptable audio quality * Take outbound calls from Asterisk to PSTN * I do NOT need a did. (no need for incomming calls) * I prefer pay as you go with little/no fixed monthly fee * They need to provide call termination to PSTN in the USA, Canada and Japan * Price maters but is secondary after the above = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USA dial plan
--- Senad Jordanovic [EMAIL PROTECTED] wrote: Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with 1 in order To successfully make a call to other USA destinations? Not always. My local phone company (Verizon in So. California) said we'd have to dail 1-310-xxx- for local numbers (I'm in 310 area) but as of current time I can still dail just 7 digits and it works fine. I think over time, yes, we are headed to the requirement to alwauys dial 1-10 digit number I have not been to USA (yet) :) Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] benevolent dictatorship, or inclusive developper community?
What hapens in other Open Source projects is that someone gets _really_ frustrated and also happens to have some free time so he copies the current code into his own CVS system and runs with it. If he does a good job user follow. I imagine that if someone were to take the current code, apply all the patches there are out there, test it with a beta program and then documet the features no one would bother using Digium's CVS system any more - they'd go for the stable tar file. Most people I think feel it is easier to put up with the current situation then to step forward the take on the job of doing it right. It would take a lot of time. I can envision someone wanting to run a company that did mostly Asterisk consulting wanting to take on this job. being the place everyone goes to get the slick-packaged Asterisk code would get them a reputation and a leg up on the other Asterisk consulting companies. But as above, It would take a lot of time. --- [EMAIL PROTECTED] wrote: sorry for the cross post, but this is germane to the developpers as well as the larger user community. Re: [SIP 104]: [patch] Cisco-like NAT trick for outbound SIP connections On Sat, Jan 03, 2004 at 08:07:29PM -0600, [EMAIL PROTECTED] wrote: A BUGNOTE has been added to this bug. tabarnac! it's been months now! the only thing that i can think at this point is that mark doesn't want sip to work through nat. i am getting very frustrated with digium's benevolent dictatorship of this project. how to make the asterisk project more inclusive of people's development efforts and contributions? is it time to start thinking about a fork? -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] benevolent dictatorship, or inclusive developper community?
-dev = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sendmail problems
You say The server crashes I assume you mean that Asterisk core dumps and sendmail continues to run just fine. If you can send mail out of the box sendmail is confgured well enough and I doubt the problem is there. If you can get Asterisk to dump then what you need to do is use a debugger to get a backtrace. This will tell to the line (as i line of coe) that caused the crash. The thing to remember to that if a program crashed it is due to t bug.. There _should_ be no way for a user through misconfiguration to cause a core dump. What you are looking for is a little bit od C cde that doesn't handle some condition well. If yu use gdb and the bt commad you can find the line Asterisk was executing when it crashed. I'd not suspect sendmail --- [EMAIL PROTECTED] wrote: Hello, I'm having some * and sendmail integration problems, probably because i don't know too much about sendmail. My server crashes when I forward voicemail from one * voicemail box to another, everything else works. E-mail notification works on all boxes when new mail arives, the problem only seems to occur during this forwarding function. It's a difficult problem to troubleshoot. If I start * -gc, the server doesn't crash, just hangs up for about 60 seconds then completes the task, so i can't seem to get a core dump to dive into the specifics of what's going on. I'm not sure how to debug sendmail to look at that side. If someone would be kind enough to e-mail me some sample sendmail.cf files, I may be able to see if I'm not configure properly. I've been reading the sendmail.org site but this application is really archain and difficult for me to understand enough to fix it myself. Thanks in advance. JR ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PBX Functionality How-to
One thing Centrex is that Asterisk is not is a turn key system. With Asterisk you have to either build the PBX your self or pay someone to build yu one. With Centrex you simply write a check. THat said, you can build anything you want so of cource the feature list can match. The best way to learn what it can do is to build a small PBX with just a couple extensions. Try to build in all the funtionality you need in your larger system. If you get into trouble you may want to ask __specific__ questions like I want to make XXX work, I triedd XXX and YYY but I still have this problem it it? You may have to post 50 questions like that one at a time. But you will get answers. Asterisk has a learning curve. expect it to take a few weeks of study But the bottom line is that Asterisk will do quite a bit more than Centrex. I don't think Centrex does VOIP at all --- Christopher J. Wolff [EMAIL PROTECTED] wrote: Hello, I had a partner of mine present a Centrex 21 brochure and ask how many of those features can I fulfill. There is nothing out of the ordinary, it's stuff like call hold, call forward, 3-way calling, etc. Has anyone assembled a how-to that shows how to configure PBX or Centrex type functionality? I found one in the voip-info wiki but only a couple of topics were filled out. Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Starting Asterisk
Look in the directory /etc/init.d (/etc/rc.d/init.d on some systems) You put a script in there called asterisk. There is a sample called asterisk.init in the source. copy it to /etc/init.d/asterisk You may want to study the other files in /etc/init.d to see how they work. Next read the chkconfig man page and see way you'd want to type chkconfig --add asterisk; chkconfig asterisl on Finally to start asterisk you can type ./asterisk start You may also want to re-boot the computer to verify that asterisk does start automatically --- [EMAIL PROTECTED] wrote: On Tue, Dec 23, 2003 at 12:18:10PM +, Adthrawn wrote: Hi, Can anybody guide me in configuring the system to start Asterisk from bootup... Probably a highly remedial question - but you've got to start somewhere! If you use screen(1), you can do screen -d -m to start asterisk, and able to reattach to to it using screen -d -r. A sample would be like screen -d -m /path/to/asterisk -vgc Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
The reason you use UDP over TCP for realtime meadia is that TCP's ability to reliably deliver every packet in order actually sounds worse. Reason being is that with a UDP system a dropped packet sounds like just a dropout but if you used TCP the audio stream would be held up and delayed in a queue while that lost packet was being retransmitted. In stead of a dropout the audio would sound as if someone kept hitts a pause button on a tape recorder. A dropout sounds better then a delay of potentialy several seconds Almost all realtime meadia systems (telephony, video, possition reporting and so on) maintain some kind of a buffer on the recieving end. But you trad the buffer lenght for delay. Using UDP allows the application to do the buffering where as TCP putting this buffing functin in the operaing systems network code. --- Andres [EMAIL PROTECTED] wrote: On Tuesday 23 December 2003 11:40, Rich Adamson wrote: There's no reassembly with udp, and there is no sense of packets arriving in the same order as what was sent. Udp is a best-effort low-overhead way Right, UDP itself does not care about order, but at the application layer you can keep track of it. You can design your application to buffer X packets and then reorder them according to sequence numbers. of transmitting data (with UDP often times referred to as the Unreliable Data Protocol). Changing to TCP would allow reassembly, however the overhead would be substantial. The problem occurs when the software is expecting the packet in a certain timeframe so that it can reassemble it in a timely manner. It's not a big deal with a web page or something along that lines. But when a voice application cannot get reassembled in a timely manner, you'll surely notice it! -Original Message- From: Joel Maslak To: [EMAIL PROTECTED] Sent: 12/23/2003 10:41 AM Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms) On Tue, 23 Dec 2003, Rich Adamson wrote: If a collision or dropped packet occurs (in a voip udp environment) there is no way to retransmit the missing/damaged packet. Missing one packet isn't a big deal, but if you have collisions and/or dropped packets, there is a very high probability that lots of packets will be dropped. If too many are dropped, you'll hear the result in the undecoded voice as choppy voice. Actually, collisions occur at Layer 2, not Layer 3, and the layer 2 hardware automatically resends packets involved in a collision - layer 3 is never aware of it happening (although it may cause additional delay). Eventually the ethernet card will give up if too many collisions occur during retries, but this is very rare in practice unless the network is *VERY* loaded. Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg ethernet would handle roughly 20-25 rtp sessions before bumping into the problem (your milage may vary). The majority of the folks on this list seem to be running home/soho systems and would likely never run into the issue. But the heavier users will. For a duplex mismatch, my experience is that if one end on a 100 Mb/sec link is half and the other is full, bandwidth is limited to about 8 Mb/sec max. This is based on some tests I've accidentally conducted. If you try to send 9 Mb/sec over that link, yes, some packets will get dropped as they simply won't fit. (But I do agree that for a half-half link, you can get about 20 Mb/sec) -- Joel ___ 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 ---End of Original Message- ___ 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Audio
My Strandstream BT100 is working OK for both inbound and outbound now except that when you speak into the handset you cannot hear your own voice in the earpeice. It works OK, the other end can hear the call but most telephone users have become used to hearing their own voice. Is this something I can fix or is it a feature of the GS phone? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting audio gain for SIP extensions?
Is there a way to set to audio gain for each SIP extension? I see in the docs this can be done for zaptel but I don't see it documented for SIP. It would be nice to be able to make the various kinds of extensions have equal volume. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
--- Darren Nickerson [EMAIL PROTECTED] wrote: Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context = any) You are lucky. I'm getting this: -- Incorrect password '1334' for user When I enter 1234. I'm using dtmfmode=rfc2833 and a GS Budgtone 100 phone. Why do I getr 4x while you get 2x ?? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
I think this is a problem on the Asterisk side. I'm seeing the same problem using a Grandstream Budgetone 100. And the GS does have setting for both in-band and RFC2833. My guess is asterisk is accepting the DTMF tone __both__ ways It is reading the RFC28833 stuff _and_ hearing the audio tones as well. --- Tilghman Lesher [EMAIL PROTECTED] wrote: On Sunday 21 December 2003 00:29, Darren Nickerson wrote: Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context snip Changing the DTMF mode would indeed seem to be the logical solution. However, it appears that SJphone does not support that option (after a quick perusal of their PDF). You might want to file a bugtracker request on their website to implement that functionality. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More beginner questions
--- Jon Creasey [EMAIL PROTECTED] wrote: Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Try this one the [fwd.pulver.com] section. Yes I see you have it in general. But fore some reason it needs to be there too disallow=all allow=ulaw allow=alaw Does anyone have any suggestions. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Headless Linux system for Asterisk
There are no issues. There is no reason to have a K/B or monitor on the server. Just sucks up power and adds to global warming. You may also want to pull any CDROM or flopy drive from the box too for the same reason. Seriusly, you should be using ssh from a remote machine to access the server. I did install X11 and many X11 clients like but NEVER run X11 on the server just do ssh -X and work on a remote machine. --- Michael Welter [EMAIL PROTECTED] wrote: Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? TIA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fedora core 1 install problem - CAN SOMEONE ELSE HELP HIM
Reinstaling the OS is not going to do anything ut get you right back to where yu are now. Look at the last command the the Makefile tried to run. Did it choke while running bison? If so run the bison command by hand. What happens? You need to cd to the directory where Makewas at when it tried to run bison first. You might also try typing which bison just to verify that bison is installed and in your path. In general when a Makefie fails you try to get the failed command to run by hand and then you fix up the Makefile to do whenever you neede to do by hand. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIPURA Breaches Contract
Does this kind of stuff really belong here? Heck for all we know you guys did something dumb to make them mad at you and they had good reason for their actions. Who knows, why should we care?? Someone here said you guys charged him _prior_ to shipping an item. If anyone charged me in advance of shipment I'd blacklist them forever. Again who knows what's up but the bottom line is one sided rants and self defeating. No one cares who's right no one was any why to know either so arguemts are pointless. (Installing e-mail filters now to prevent flame war...) --- John Brown (CV) [EMAIL PROTECTED] wrote: Hi list, Well I really didn't want to see things get to this point, but Sherman at Sipura along with their President Jan F. leave me no other choice. SIPURA has been provided a letter from our attorney for Breach of Contract and damages. They have yet to respond. A quick background. 1. Sherman (SIPURA's Director of Marketing), stated that we would do a join press release for the Oct VoIP conference in Long Beach. The day the release was suppose to go out, he decided not to do it. We had agreed to pay of half of the cost to get the release out. 2. SIPURA and Chagres negotiated a contract, where Chagres was to provide logistical support for sample sales of the SPA-2000. This contract was executed by SIPURA's President Jan F. One day after the contract was mutually executed, Sherman from Sipura, stated that instead of having us do the logistical support they would handle this within the company. The following day, I was advised by several ASTERISK list members that SIPURA had a company listed on their website doing order processing. They asked if this was the new company we where forming to handle the VoIP hardware. Upon review of both SIPURA's site and the other company it was clear that SIPURA had sourced this service to another firm, AFTER IT EXECUTED A LEGAL AND BINDING contract with us. Chagres spent considerable time and money putting into place the requirements outlined in the contract. Those efforts are now lost money 3. In conversations with SIPURA's President Jan F. he committed to having product available for us to pickup in Hong Kong on 1-Nov-03. When we contacted Sherman on 4-Nov to confirm pickup, Sherman would not commit to the order, pickup or anything else. All he wanted was for us to pay them the money for the equipment and then we can talk. SIPURA's President Jan F. has not returned any further phone calls. 4. Sherman had agreed to provide us with leads for 50 some odd different companies that where looking for SPA-2000 product. When we asked him again for those leads, he refused to provide them. In general, I beleive that the actions of SIPURA and its managment team have been less than honorable, they have damaged Chagres and have hurt our customers. Sipura has a simple and clear path to resolve this matter. Honor the contract they signed with us. SIPURA has also further breached their agreement with us by removing us from their Partners page. I remain hopeful that SIPURA ownership and management will see the honorable course of action and live to the terms of the contract we executed. Should they decide to continue to ignore us and our attorney's attemts to communicate we will take legal action in court. Sipura has a GREAT product, its sad to see that their management team is willing to breach contracts. To our customers that have pending orders. We have attempted to source inventory from another location and have not been able to do that. There for we will be issueing complete refunds for all SPA-2000 purchases. While we have been difficult to reach in November, I remain strong in the belief that we will do the right thing by our customers, and that our ability to service further (starting today) orders has vastly improved. Cheers John Brown ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Telemarketer Torture
Very good. I made a much smaller one that just goes in an endless loop, no way out but to hang up. I figured telemarketer are too stupid to notice the same prompts over and over. I might use yours. Did you put the .gsm recording some place we can get them? My brother has the BEST solution for sales people. He makes an appointment with them to come out and gives an address across the street. It really wastes a real estate salesman or house painter's time to drive out to a dead end. Keeps em off the phone too. --- Steve Murphy [EMAIL PROTECTED] wrote: Hello-- I submitted of extensions.conf that contains my telemarketer torture menus, last week sometime to the mailing list. I got back a note from the mailing list machinery, stating that it was too big, and would be subject to approval. No such approval came, I guess. Either I missed it, or it didn't rate, or the moderator just plain hasn't gotten around to it yet. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sendmail not on localhost
You would have to look at the code in the VM app. and see if the hostname for the mail server is configurable. Likely it is simply hard coded to localhost which would send the mail to port 25 on the * sever. In theory the VM application _could_ use a remote mail server but it would have to be written that way. I'd prefer to run a local sendmail. Ths means you have a local queue and the mail gets handed off quikly even if your other server is down or slow. --- Ralf Illing [EMAIL PROTECTED] wrote: Hi . I already set-up sendmail on another network server thus it would be nice to use that one or is sendmail on * server required!? I had a look in the archive but couldn't find any information where to set the mail server from localhost to my network server . Cheers Ralf = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] next stable release?
I think the word here is that stable release is a do it yourself project. You need to set up a test environment where you can check out new CVS updates, apply patches you want and see that it all works. Only then would you but the code on a productin system. --- Brancaleoni Matteo [EMAIL PROTECTED] wrote: you should not fear cvs. many of us are using current (or semi-current) cvs version in production systems without issues. if you're in a test environment, you won't have problem. Also many of latest cvs additions are bug fixes, nothing really new, apart of cdr_odbc. See asterisk-cvs list more more details on that. matteo Il mer, 2003-12-10 alle 18:47, john lawler ha scritto: Hi guys, I've been running 0.5.0, which is dated sometime in September of this year and I've noticed a couple of new features in more recent code that I'd like to use, but am hesitant to go w/ CVS code. My system is not exactly a production system, it's mostly test, but I'm still leery of the fresh code. I'm wondering when the next stable release might come out, and how those work in general. Thanks, jl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
I've set up a test enviroment and beed trying to answer that question. I think if you are carfull NOT to do dumb things like running X11 and a browser and so on on the server you can use a pretty low power system. Just do not plug in a CRT, mouse or keyboard. Use telnet or ssh. The requirements to run a graphic interfaceare are greater then to run a low-end asterisk server. Asterisk seemed to run well on am old 400Mhz Pentium but I'm using an ADM2600+ with 128MB ram and am not taxing the system much at all. I think a 1Ghz Pantium would be well more then required. OK that said. BIG remaining question. I've got some echo problems with the FXO card. Fixing this might take a lot of CPU power to do the required DSP. I don't know yet. But it works with two calls open at about 2% of the CPU utilization. ond the ADM 2600+ Pushing 8K sample/sec data aound is a very lightload audo at 8K is a very low data rate. My goal is to reduce the heat and electic power. I may try _under_ clocking the 2600+ and see if that makes it run cool enough that I can remove a fan. --- Trench Shoring [EMAIL PROTECTED] wrote: I have been reading asterisks and everything I can get my hands on for the past week. I want to know what class processor is the bare minimum I need for a four port Asterisk installation? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
We assume this kind of experimentation is happing in a lab enviroment. Failure there is not a problem it's a data point. In a test system I can take out half the RAM, slow the CPU clock or run the CPU without the cooling fan and just measure what happens. Yes, stupid do do those things in a system people are depending on. --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: There is a _good_ reason to ask too. I've been experimenting I buy new equipment but I'm still looking to reduce power, heat, noise and space to the bare minimum. No need to buy a CPU that burns 120W of power if you can use a one that uses 45W and lets you get rid of one of the fans. Same with disk drives. More RAM might let you use a low noise/low heat drive rather then that 7200RM noise maker. I'd like to be able to install a notebook sized drive on the * server. No, actually that's a terrible reason to ask. If you are unfamiliar with * you have no business trying to optimize your system. This is typical early optimization that plagues any design or deployment. Just learning * is NOT the place to try and cut corners and save some coin. Get it working, get familiar with it and THEN see where your specific needs are and optimize for them. Early optimization is a problem everywhere, not just in programming. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk servers sharing/propagating registry ?
It would be a major change to the code but I think what you'd want to do is have the Asterisk server store _all_ of it's information in something like a database, The dail plan, SIP registrations, everything would have to go there. Once you've done that any number of Asterisk servers could share the same database and there are methods of running mirrored databases already. When I worked at a dot.com we had a design requirement that I should be able to go into the server room and pull any of the AC power cords and the users should not be able to know. About the _only_ way to do this is with load balancing. Fail over does not work so transparently. With SIP or other VOIP phones there is a chance of doing a kind of load ballance but with an analog phone wired to a channle bank, if the CB smokes Good News: as far as I can tell Asterisk already does keep much of it's data in a Barcley style DB does it not? If so Asterisk is 3/4 the way there. But I think the market for this is someone with 5000 extensions who needs five nines of reliability and Asterisk has other things to do before it can be used for such a system. mainly getting dail plan info out of those .conf files. --- Florian Overkamp [EMAIL PROTECTED] wrote: At 22:47 8-12-2003 +0100, you wrote: The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com (thus ending on a random asterisk servers) - but after that, all the servers would be aware of the registration. Thus any asterisk server would know how to route a call to SIP/some registered user Same thing for IAX peers. Of course, setting up various IAX links between each server is no problem (with registration cascading, for instance). Registration cascading is not possible (I think) but could it be solved with a shared dial route: Instead of DIAL(IAX/sip.isp.com) could you not DIAL(IAX/sip1.isp.comIAX/sip2.isp.comIAX/sip3.isp.com) to reach a similar effect ? (or chain them in different lines so it tries to reach the first one, then the second one if it fails, and the third if that fails. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO cards - low cost systems
Comments on this are welcome. Here is my opinion... I just went through this. Your office size is not economical. Actually smaller or larger would be better. Getting a channel bank and then using only 8 ports is a waste. OK if you have 24 extensions but 3x to expensive if you only use 8 ports. I'm working on a 3x8 system for a not-for-profet I'm doing the work for free and they value every cent as they live off donated money. (You don't want to tell donors that 100 of their $20.00 checks went into a new PBX and did nothig to save sick homeless kids or whatever.) Here is what I told them: Go VOIP. You can get decent IP phones like BT100, or whatever for $65 to $100 each. Buy eight of these. Next get a VOIP service provider to provide you with a PSTN DID (A phone number) VoicePulse will do this for about $8.00/month pluss outgoing per minute cost. So you get as many incomming lines as you need and you have zero hardware interface at your site. (other then your DSL line.) Keep one or two analog POTS lines and use one or two FXO cards either Digium other. You want POTs for free local calls and for 911 calls and for if the VIOP service fails or if the DSL line is down. But one POTS line for the whole office is enough So now your Asterisk server is just one re-cycled PC with one FXO card installed, nothing else. Pretty dard cheap. If using a re-cycled computer as the server _do_ keep spares on site. You don't want a broken power supply fan to crash your phone system for a day. I'm suggesting a full-up hot spare Asterisk server system be kept on-site. --- Michael Rowley [EMAIL PROTECTED] wrote: Hey, Here is a quesion for you. I am still battling with the phone system for my new buisiness. 6 incoming lines, 1 fax, DSL. 8 phones max, will provably start with 5 to save money. I was thinking of using Asterisk, but having difficulty finding appropriate buisiness phones. The Mitel 5055 is the best one I have found, but the price seems to be about 400$ per phone. $2K, plus a 500$ server, then how to get the 6 B1(pots) lines into it. I had thought of using a channel bank, but what a pain in the ass that is becoming. For one, they are expensive, and I then have to buy the T1 card for the phone server. I though, why not go with an FXO card. I wish there was an X400P card with 4 ports on it, but, que sera. I can get them for 100$ apiece, or $50 for the knock offs on ebay, but that means 6 pci slots. Not easy, I could use one of the pci extender boxes, but now I am worried about conflicts. Or dialogic analog 4 or 12 port cards for about 1500 to 1800$. :( This is getting expensive. Part of the idea was to save some money. The other part was to use open software as much as possible, and support the FOSS community where ever possible. Here comes the question, wait for it :) Has anyone had success with the dialogic 4 port cards, running 2 of them in a server with * in a buisiness environment as stated above. I am begining to think that I may be better off just going with a proprietary system and cough up the 6K and get it over with. There are a couple of solutions that will share the cat5 cable, that's something. Any ideas? Suggestions? Does anyone know of a solution provider out there who will be able to set this up for me for the 6K the phone system is going to cost me anyway? Michael Rowley MD FP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
