[asterisk-users] Compatibility INTEL E7520
Hello,I am buying a server for my Asterisk PBX (with 2 Digium TE110P):- IBM x346: (2005-2006) ChipSet: INTEL E7520Bus: 800 MHz.Proccessor Support: 2.8, 3.0, 3.2, 3.4 , 3.6 y 3.8 GHz, del tipoExteneded Memory 64 Technology (EM64T)I read the Digium Compatibility List and there isn't among incompatible chipset models.Does Anybody uses that chipset (or machine) with Digium cards successfully? Thanks in advanceDaniel ___ --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] About IVR and Oracle
maybe you could make it with PHP and its driver for Oracle. Daniel Pizarro www.infobox-peru.com-- Forwarded message --From: Moises Silva [EMAIL PROTECTED]Date: 23-ago-2006 17:12Subject: Re: [asterisk-users] About IVR and OracleTo: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSure is possible. Look into google 'asterisk agi fastagi'.RegardsOn 8/23/06, Javier Lara Sanchez [EMAIL PROTECTED] wrote: Dear All, I need to buid an IVR that could make a request to a data base (oracle) in a remote host. The idea is that an user dial a extension with 2 options and one of them ask for a data (in the case a date). This data is the field that the data base needs to find the information that the user are looking for.. Somebody know if this is posible or have any idea where can I find information about this? Thank Regard Javier ___ --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 --Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Asterisk Users Group - Portugal
Boa noite, Eu também estou procurando criar uma comunidade de Asterisk no Perú para aproveitar das experencías de todos. Se vocês criam a sua comunidade, por favor contem con a minha ajuda e colaboração. A minha pagina é http://www.asterisk-peru.com Saudações, Daniel Pizarro Lima - Perú2006/6/23, Marco Mouta [EMAIL PROTECTED]: Oláa todos! :)Hi all,Thanks for all of your fast replies, further now may be simpler totalk in english. But any one could write in Portguese, any way theimportant is to make it happen.This weekend i will post here some topics could be interesting: mailing list and a blog or php forum don't know, i must say i'm not aweb expert.I work with VoiceXML VoIP more related to communications.Mailing list and blog or forum seems easy to start this, share and learn. I hope i can help to this project grow.Best regards,Marco MoutaOn 6/23/06, Josué Conti [EMAIL PROTECTED] wrote: Marco, boa tarde (in Brazil) Na realidade tenho interesse em estar participando deste sua idéia também, posso contribuir no que for necessário para que avance a idéia. Cumprimentos Josué I have interest in participating of this its idea , I can contribute in that will be necessary so that it advances the idea. Best Regards. 2006/6/23, Marco Mouta [EMAIL PROTECTED]: Boa tarde, Após alguma experiência com o Asterisk, e com muito ainda para aprender, gostaria de saber se há alguém nesta mailing list que pretenda criar um Asterisk Users Group para Portugal. Visto que acaba sempre por ser uma enorme aprendizagem ( valor acrescentado) a partilha de experiências/problemas e soluções nas implementações Asterisk. Há spre detalhes que variam entre os Telco's de cada país, voice prompts, etc. Se houver um número minimo de pessoas interessadas, podemos avançar com a ideia. -- Com os melhores cumprimentos, Marco Mouta ___ --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 ___ --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 --Com os melhores cumprimentos,Marco Mouta___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] How to redirect an incoming call to an external phone numer
Of course... you only need to Dial to other port FXO connected to PSTN and passing the number as extension:
[redirection] ; your inconming-calls context
exten=s,1,Dial(Zap/${OTHER_FXO}/${EXTERNAL_NUMBER})
exten=s,2,Hangup
2006/6/1, Javier Rodriguez [EMAIL PROTECTED]:
Hi,I would like to know if it is possible to redirect an incoming call toan external phone number. Can this be done easily?Thanks in advance,Javier___
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Re: [Asterisk-Users] Sip gateway don´t hangs up
I have the same problem. Help! On 5/30/06, Danko Miocevic [EMAIL PROTECTED] wrote: Hello, I have a SIP gateway connected to my asterisk but the thing is that when I try to use my analog phone line and I hung up the gateway doesn´t hungs up the line. I think that when I call my gateway from a softphone to get access to the analog line and I close the communication the asterisk server sends a tone to the gateway and it doesn´t understand it. I want to know wich tone is it because I can configure it from the gateway. If anyone has an idea... I´d love to hear it... thanks for your time, Danko ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Asterisk IVR / Scalability
Hello, Aheeva offers a complete Call Center solution including IVR. I read in the wiki-info about a large scale deployment in Portugal (40 E1, I guess) made by an american company. I am programming a php-ivr library, but I think it takes me some time. DanielOn 4/27/06, Dovid Bender [EMAIL PROTECTED] wrote: i am looking for a good ivr system for my company. these are my question are there any good ivr's that can be easily integrated with asterisk ? and are there anylarge scale deployment of asterisk to date ?Lots of people are using asterisk in a productionenviroment.When you say IVR system what do you mean ? A GUI thatwill help you create it ? You can create your own IVR as long as you learn how to write a proper dial plan.__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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] I am looking for a webphone on MY SITE
Well, it runs on Internet Explorer because it use ActiveX technology. Maybe somebody could build a Java applet version. DanielOn 4/26/06, Ronald Wiplinger [EMAIL PROTECTED] wrote: Bruce Reeves wrote: The only one I have heard of is WebIAX http://www.voip-info.org/wiki/view/WebIAXThis one looks ok, with the exception that it is only working with Internet Explorer!!!byeRonald Wiplinger___--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 ___ --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] problem with TE205
Why do you comment these lines: ;channel=31-45 ;channel=47-61 and put those in the zapata.conf? bchan=31-45,47-60 dchan=46 ___ --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] te110p and interrupts
use lspci -vb for detecting interrupt conflicts..On 4/10/06, Anton Krall [EMAIL PROTECTED] wrote:Guys. I have an issue with a te110p card and also some tdm04b cards on the same system:Zttest returns this for the tdm04b cards:[EMAIL PROTECTED] ~]# /usr/src/zaptel-1.2.4/zttest 38 -vOpened pseudo zap interface, measuring accuracy...8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8223 sample intervals 99.621582%8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8400 sample intervals 97.460938%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8438 sample intervals 96.997070%--- Results after 15 passes ---Best: 100.00 -- Worst: 96.997070 -- Average: 99.601237And for the te110p:[EMAIL PROTECTED] ~]# /usr/src/zaptel-1.2.4/zttest 1 -v Opened pseudo zap interface, measuring accuracy...8192 samples in 8239 sample intervals 99.426270%8192 samples in 8200 sample intervals 99.902344%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8384 sample intervals 97.656250%8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8191 sample intervals 99.987793%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00%8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793%8192 samples in 8192 sample intervals 100.00%8192 samples in 8193 sample intervals 99.987793%--- Results after 20 passes ---Best: 100.00 -- Worst: 97.656250 -- Average: 99.843140Im getting problems receiving faxes which can be caused by this.As you can see, the cards are not sharing IRQs as far as I cant tell CPU0 CPU10: 17697848 17714488IO-APIC-edgetimer 7:22IO-APIC-edgeparport08:10IO-APIC-edgertc9:01 IO-APIC-levelacpi 14: 103455 112606IO-APIC-edgeide0 15:00IO-APIC-edgelibata169:00 IO-APIC-leveluhci_hcd:usb2185:2793868 12 IO-APIC-leveleth0201:00 IO-APIC-levelehci_hcd:usb1209:00 IO-APIC-leveluhci_hcd:usb3217: 19672684 15531101 IO-APIC-levelwcte11xp225: 17824822 17371238 IO-APIC-levelwctdm233: 15323598 19896224 IO-APIC-levelwctdmSo, can anybody point me in the right direction in order to debug the problem with zttest and quality of sound?How can I change prioritiess for the cards or maybe adjust or see if theyget their right number of interrupts, etc?Thx for the help guys!___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Any Aheeva Users?
Hello, what is your experience with Aheeva? Are you a vendor or the final user? Thanks, Daniel PizarroOn 4/10/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:we'reusing aheeva on a call center on Dominican Republic, we're using aheeva since 2 month,Wai Wu wrote:Isn't aheeva a commercial product? Whoever wants to find out how it is should ask aheeva for referrals, and I recommand him personally pay visits to their customers on their expenses if he is a prospect.From: [EMAIL PROTECTED] on behalf of Kevin P. FlemingSent: Tue 4/4/2006 6:40 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Any Aheeva Users?Steve Totaro wrote: Just looking for unsolicited thoughts on the Aheeva product?Anyonehave anything to say?H soliciting unsolicited thoughts. Interesting :-) ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] asterisk box as a voip gateway
Your zaptel is wrong... it must be: zaptel.conf: span=1,1,0,ccs,hdb3 dchan=16 bchan=1-15,17-31On 4/7/06, JP Carballo [EMAIL PROTECTED] wrote: Mark Quitoriano wrote: Hi Guys, Im configuring my asterisk box as a voip gateway. I have TE110P which is connected on my PBX and i will be using voip for my outgoing. Here's my config zaptel.conf: span=1,1,0,ccs,hdb3 fxoks=1-32 zapata.conf: context=default signalling=fxs_ks group=1 channel =1-32 -- Regards, Mark Quitoriano, CCNAWhat seems to be the problem?--JP Carballohttp://www.netfone2x.comBringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
[Asterisk-Users] Re: E1 te110p problem
Hi, What kind of problem happens? Show your dialplan. Daniel On 4/4/06, Toke [EMAIL PROTECTED] wrote: Hi Antonio, What problems are you having with it? Which operator give you E1 connectivity?? If you want mail me directly and we will try to have it working if it is possible. Regards On 4/4/06 10:38, Antonio Almodóvar [EMAIL PROTECTED] wrote: Hi all. I'm using a te110p in spain. ;zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 I'm getting problems dialing out through this span. ¿How can I debug its behaviour? Thank you in advance. ___ --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 ___ --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] TDM400P busy
Maybe your lines use polarity reversals for hangup detection.On 3/27/06, Il Neofita [EMAIL PROTECTED] wrote:Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it takes the line occupied. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] problem with tdm22b
Diego: I have a tdm22b and the first two ports are FXS and the next two FXO. So the zaptel.conf must be as: fxoks=1,2 fxsks=3,4 Saludos Daniel Pizarro Infobox Peru ex alumno de la PUCP On 3/17/06, Diego Quintana Cruz [EMAIL PROTECTED] wrote: Hi everyone,I have a problem installing interface card tdm22b in a debian etch machine.First I added manually the zaptel module:apt-get install zaptel-source kernel-headers-`uname -r`m-a a-i zaptel Then I dotumiwall:/usr/src# ztcfg -vvZaptel Configuration==Channel map:Channel 01: FXS Kewlstart (Default) (Slaves: 01)Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03)Channel 04: FXS Kewlstart (Default) (Slaves: 04)4 channels configured.ZT_CHANCONFIG failed on channel 1: No such device or address (6)Does anybody knows why zaptel is not detecting the device?? My /etc/zaptel.conf contains:fxsks=1,4fxoks=2,3which I think is the correct order of my fxs and fxoRegards from Peru,--Diego Quintana a.k.a. RouterMaNEstudiante Ing de las Telecomunicaciones PUCPLinux Registered User #382615 - http://counter.li.org/http://routerman.blogsome.com http://planeta.debianperu.org___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Problem with two cards Digium
I think you need to check your interrupts. You can use: cat /proc/interrupts and check if the modules have not been shared with other devices. Also, the command lspci -vb shows it detaily. This kind of conflict was a nightmare when I tried to configure 3 board (TE110P, TDM2406, TDM2402). I had to try all the pci slots until I have every module in its own interrupt. My zaptel.conf looks so: span = 1, 1, 0, ccs, hdb3, crc4 bchan = 1-15, 17-31 dchan = 16 fxoks = 32-79 #declare all 48 ports ... in the zapata.conf you 'll define only the valid ones This is the order of loading for me: modprobe zaptel modprobe wcte11xp modprobe wctsm24xxp Oh! But first I run a script on startup to unload the modules because for some reason, it doesn't work when the linux starts. I hope it helps you. Daniel Pizarro On 3/2/06, Saul Diaz [EMAIL PROTECTED] wrote: Bart Fisher wrote: I'll guess the TE410P is being loaded first - Try swapping entries in zaptel.conf and zapata.conf Bart - Original Message - From: Bartosz Supczinski [EMAIL PROTECTED] To: Asterisk-Users asterisk-users@lists.digium.com; Asterisk-Dev asterisk-dev@lists.digium.com Sent: Tuesday, February 28, 2006 6:57 PM Subject: [Asterisk-Users] Problem with two cards Digium Hello, I`ve got a problem which I can`t deal with. I own 2 cards - TDM2400P and TE410P. I`ve put them into a HP Proliant DL380 G4 server, compiled the drivers according to the manual. Unfortunetly there are both cards channels are configured in zaptel.conf file the first module (in this file) sends an error. For configuration: span = 1, 1, 0, ccs, hdb3, crc4 fxsks = 21-24 bchan = 25-39, 41-55 dchan = 40 root# modprobe zaptel root# modprobe wctdm24xxp -- ZT_CHANCONFIG failed on channel 25: No such device or address (6) FATAL: Error running install command for wctdm24xxp For configuration: span = 1, 1, 0, ccs, hdb3, crc4 bchan = 1-15, 17-31 dchan = 16 fxsks = 145-148 root# modprobe zaptel root# modprobe wct4xxp -- ZT_CHANCONFIG failed on channel 145: No such device or address (6) FATAL: Error running install command for wct4xxp If the configuration applies only a single card the modules are loaded correctly. The analog card is equiped with one FXO module, which is placed in the last joint. I`ve disabled hyperthreading in my kernel and in BIOS, interrupts are not shared. Besides I`ve put the cards in other slots, switched on and off ACPI and other functions in BIOS as well as in my kernel. Maybe the problem is in the drivers? I`ve attached some information which might be useful. http://www.dir.pl/~supczinskib/logs.tgz -- With best regards Bartosz Supczinski IT Manager DIR Konstytucji 3 Maja 2 86-300 Grudziadz POLAND www.dir.pl t.: +48 (56) 6440100 f.: +48 (56) 6440111 m.: +48 (504) 019040 ___ --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 ___ --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 Check dmesg toojust in case..___--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 ___ --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] How to determine the power draw on TDM2400P?
I am using both TDM2406 and TDM2402 and using true 550W power supply. I couldn't find any indication from Digium about power comsuption.Daniel PizarroOn 2/28/06, Melisa Teoh [EMAIL PROTECTED] wrote: Kevin P. Fleming wrote:Melisa Teoh wrote:For example, how much power would a server with 3 of the TDM2400P fullyloaded cards draw?With what type of modules? The power consumption figures are on the TDM2400P data sheet and on the Digium website, IIRC.TDM2451E x 3 (60 extensions; 12 lines)Did you mean this datasheet? http://www.digium.com/downloads/product_sheets/tdm2400p.pdfTheredoesn't seem to be an indication of power draw.Thanks.Melisa.The only significant power draw is the FXS modules, and then only when actively ringing. The remainder of the board is powered via the normalPCI bus power connections.We're trying to figure out what APC power we should have to achieveapprox 30 minutes back-up time. Well, the boards cannot draw more power than your server can supply, soif your server has a 600W supply (for example), use that for yourcalculations.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
[Asterisk-Users] Generating Polarity Reversal on FXS line
Hello list, Is it possible to send polarity reversal signalling on FXS lines in Asterisk using the TDM240X cards? I need to send that signalling to legacy PABX using TDM boards of my Asterisk box. The unique clue I could find was: http://lists.digium.com/pipermail/asterisk-users/2004-December/thread.html#77646 Thanks in advance, Daniel Pizarro ___ --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
