Re: [Asterisk-Users] Language Problems

On Sat, 26 Feb 2005, Anton Krall wrote: > and then Spanish as > /var/log/asterisk/sounds/sp > /var/log/asterisk/sounds/sp/phonetic > /var/log/asterisk/sounds/sp/digit > /var/log/asterisk/sounds/sp/letters > > Now, the normal voices ARE heard in spanish but all digit related voices are > taken from

[Asterisk-Users] Polycpm SP300 problems

Hi, all I am trying to connect Polycom 300 to Astersik. I do not want to use FTP server for now, so I am tryng to set up phone manually. Network configuration parts is OK, except that it does not ask for SIP server address. Any ideas where to set this? Also i have some problems with setting up a

[Asterisk-Users] Language Problems

Guys Im having a few issues with Languages. Ive setup the english language is it came from default: /var/log/asterisk/sounds /var/log/asterisk/sounds/phonetic /var/log/asterisk/sounds/digit /var/log/asterisk/sounds/letters and then Spanish as /var/log/asterisk/sounds/sp /var/log/asterisk/sou

[Asterisk-Users] FRS & *: an actual business use

I've noticed a growing number of stores using FRS radios. It would make sense to interface (via soundcard/console driver, with the nessacary electrical conversion) a VOX FRS radio to asterisk to allow someone in the office to page/talk with people on the floor or warehouse. You could throw that

Re: [Asterisk-Users] Re: FRS and GMRS via *

Rich Adamson wrote: GMRS, FRS and MURS radios may not be interconnected with the PSTN (47 CFR 95.141). There has been a lot of talk from lobbyists to clarify this rule, but as it stands you could conceivably connect a *private* network to GMRS or MURS radios (you can't make any plugins or modifi

[Asterisk-Users] playing "i" invalid context to an internal user

When the call comes from outside on a certain context to play "i" invalid extension to an external user is easy just by enclosing in an incoming context: exten => i,1,Answer exten => i,2,Playback(pbx-invalid) How to play an this context to an internal user, internal user has access to all contexts

RE: [Asterisk-Users] Transfer a call ? Am I looking fortheflashcommand ?

[EMAIL PROTECTED] wrote: > Hello Jim, > > thx for the answer.. > Im happy I found someone that is using flash :) It's not perfect, but it can be useful. > Am I right, if I transfer a call with flash, the line will be free > afterwards ? Yep > Would you mind to past me how you did the flash p

Re: [Asterisk-Users] Asterisk With Broadvoice

Christopher -- regarding the country checking: does your AGI also check for mobile/cell numbers? Checking for just country codes is trivial and I do it in the dial plan, but knowing which number is actually a mobile call is tougher as each country does it differently. So I'd be interested in a solu

[Asterisk-Users] Seting up for afirst time -- can not call

Hi, all I am setting up Asterisk for the first time and have some problems. Setup is very simple -- Astersik box and two Polycom SP300 phones. I will add bells and whistles as I go, at the moment things are very simple. No TFTP servers, so phones run with their default configuration. I set up IP

Re: [Asterisk-Users] Asterisk With Broadvoice

I have mine set up where you just dial 6 to get out and if it is busy it rolls over to the next avaliable in the trunk. - Original Message - From: "Greg Hill" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, February 25, 2005 7:46 PM Subjec

Re: [Asterisk-Users] Asterisk With Broadvoice

Here are the official instructions from broadvoice for setup of Asterisk. Other configurations are not supported. http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] Asterisk + SER

I just installed SER last night but if you want it ot talk to Asterisk I found that you should install FREERADIUS Server and RADIUS CLIENT. For it to function properly - Original Message - From: "Nitesh Divecha" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Disc

[Asterisk-Users] open 723

has any one implemented open 723 at http://www.readytechnology.co.uk/open/g723.1   ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http:

[Asterisk-Users] Asterisk with regular analog phones

Can regular analog phones be used and act as extensions, or does an fxs device need to be put into place. I saw this on voip-info.org. How would extension setup be possible without the fxo being aware of the name of the device? for Analogue Phones connect to Zaptel

Re: [Asterisk-Users] HELP NEEDED ASTERISK AND MEDIATRIX 1102

Do yourself a favor and get a Sipura SPA-2100 - much easier to configure and the quality is better than the Mediatrix unit. First of all - do you have the Mediatrix Unit Manager software? If not, configuration will be nearly impossible. Secondly, you will need to configure the sip ports on the m

[Asterisk-Users] Asterisk + SER

Hello All, Has anyone tried Asterisk with SER.? My main focus is billing and authentication of my endpoints. I want Asterisk to handle all my endpoints and SER to do billing/accounting stuff. Any help will be highly appreciated. Neel ___ Asterisk-U

Re: [Asterisk-Users] weather asterisk@home

My perl is not that great, but from your debug output, the "STREAM FILE" agi command never executed from the festival-weather-script.pl script, which should have happened right away. Does your text2wave work? Is there a tts-.wav file in your var/lib/asterisk/sounds/tts dir? a tts-???.txt fil

Re: [Asterisk-Users] Asterisk With Broadvoice

On Fri, 25 Feb 2005, James Taylor wrote: > I have two Broadvoice "lines" and there's three people in the office. > Any way to: > > 1) "Pool" the connections for "trunking", where any one can get a "free" > line? > 2) Prevent more than 1 simultaneous call per "line"? (So I will not get > hit for 3.

Re: [Asterisk-Users] weather asterisk@home

Still no weather... AGI Debugging Enabled -- Executing Answer("SIP/3000-51a3", "") in new stack -- Executing AGI("SIP/3000-51a3", "weather.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi AGI Tx >> agi_request: weather.agi AGI Tx >> agi_channel: SIP/3000-5

Re: [Asterisk-Users] weather asterisk@home

Turn on debugging (agi debug) and check to see if festival is exiting with an error? (Maybe) Ernie On Feb 25, 2005, at 4:29 PM, James Taylor wrote: I'm still having problems. Festival works from command line and I can make the speakers talk. But when I dial my weather extension: -- Executing Ans

Re: [Asterisk-Users] Re: FRS radios on *

> >>GMRS, FRS and MURS radios may not be interconnected with the PSTN (47 > > > CFR 95.141). There has been a lot of talk from lobbyists to clarify this > > > rule, but as it stands you could conceivably connect a *private* network > > > to GMRS or MURS radios (you can't make any plugins or modifi

RE: [Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phoneproblem

Hello Mark , C. & All , Is this device available for sale in the US ? All the digging I've only found outside US mentions of sales . Any help appreciated . JimL On Fri, 25 Feb 2005, Mark Elkins wrote: On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have t

RE: [Asterisk-Users] Festival - Asterisk@home

Title: Festival - [EMAIL PROTECTED] Wiley, if you follow the instructions as listed it will work.   Can you post more information about what actually isn’t working? can you post the output of your cli.     Cheers, Dean     From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] O

Re: [Asterisk-Users] VoIP/Asterisk presentation

Ronald Hartmann wrote: > Anyone able to get to these I am unable to get to them. Seems I have an issue with a dead name server (the box itself) and I'll be going in and hitting a button in the next 20 mins or so, but there's 3 other name servers so I don't know why dns doesn't just jump to ano

[Asterisk-Users] weather asterisk@home

I'm still having problems. Festival works from command line and I can make the speakers talk. But when I dial my weather extension: -- Executing Answer("SIP/3000-a844", "") in new stack -- Executing AGI("SIP/3000-a844", "weather.agi") in new stack -- Launched AGI Script /var/lib/asterisk/a

[Asterisk-Users] Re: FRS radios on *

Rich Adamson writes >>GMRS, FRS and MURS radios may not be interconnected with the PSTN (47 > CFR 95.141). There has been a lot of talk from lobbyists to clarify this > rule, but as it stands you could conceivably connect a *private* network > to GMRS or MURS radios (you can't make any plugins or m

Re: [Asterisk-Users] VM+Realtime config

> 1) when Asterisk try to build the mailbox directory under the path : > /var/spool/asterisk/... Don't know about realtime, but in "standard" version, the directory is built the first time you leave a message to this mailbox hth ___ Asterisk-Users maili

[Asterisk-Users] SER vs. Asterisk - call in progress to PSTN

We're having a problem with Asterisk when we try to pass a call off to a Lucent PSTN using SIP. This behavior does not exist with SER: With Asterisk An ISDN call is started, at the T1 level we receive “call proceeding” and immediately we receive a “Call in Progress” just like the far end party

Re: [Asterisk-Users] Asterisk With Broadvoice

I have two Broadvoice "lines" and there's three people in the office. Any way to: 1) "Pool" the connections for "trunking", where any one can get a "free" line? 2) Prevent more than 1 simultaneous call per "line"? (So I will not get hit for 3.9 cents a minute. I'd like to use the country code

RE: [Asterisk-Users] "click to dial extension number" functionality ?

http://www.voip-info.org/wiki-Microappliances+SIP+Active-X+Client   http://www.microappliances.com/site/html/index.php?section=Products&page=clienthowto.php   I haven’t tried it though, let me know how it goes.       From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behal

[Asterisk-Users] CallerID Name and Digium TE405P

Hello all, I am looking at replacing our current Cisco PRI gateway with a new server with a TE405P card.  My primary concern is receiving CallerID Name info on the D-Channel.  Does anyone have any experience terminating a local Qwest PRI from a 5ES switch into the TE405P or similar?  We are

[Asterisk-Users] VM+Realtime config

Hi ALL;   When I insert data to Vm table in Realtime config, I canot see any directory built under : /var/spool/asterisk   1) when Asterisk try to build the mailbox directory under the path : /var/spool/asterisk/...   2) Where is source code that tells Asterisk to build that directory under:

Re: [Asterisk-Users] Directory config...

Right on! Have a good week-end! Francois >> How do I config Asterisk so when the directory cmd is used, the name of >> the found entry comes from a pre-record gsm file instead of being >> spelled >> letter by letter? > If the user as recorded is name, this file will be used. When it's not > reco

Re: [Asterisk-Users] How does the g.729 registration program work?

Thanks for the clarification. In that case the following should only be considered for development. Steven Critchfield wrote: On Fri, 2005-02-25 at 09:25 -0800, Mik Cheez wrote: There is an open source version of the license: You

[Asterisk-Users] Asterisk in front of Toshiba CTX

I have googled, and wiki’ed until blue. Is it possible to put T1---*Toshiba CTX ? I have a TE405P, with one interface programmed for the T1, I am not sure how to program the 2nd port to mimick the T1 to the Toshiba. The Zapata.conf   [channels] switchtype=national context=from-pstn

[Asterisk-Users] RE:Avaya Partner ACS3 and Asterisk

-Original Message- I have an avaya partner ACS r3 system that I want to be able to hook asterisk into with a x100p card, into and use asterisk to tie into a voip provider then be able to dial (or connect) to an extension like an intercom function and be able to dial a number like that. I

Re: [Asterisk-Users] Asterisk With Broadvoice

Apparently the combination of the correct registry string and insecure=very fixed it. Just as you said. Thanks! Roger Hanson wrote: - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, February 25, 2005 9:20 AM Subje

Re: [Asterisk-Users] Asterisk With Broadvoice

- Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, February 25, 2005 9:20 AM Subject: Re: [Asterisk-Users] Asterisk With Broadvoice Great! It works now!! Thanks so much. What was it that made it work? Share the

Re: [Asterisk-Users] SetCIDNum using SIP?

He was talking about SIP On Fri, 25 Feb 2005 16:09:27 -0500, Kevin Collins <[EMAIL PROTECTED]> wrote: > Nabeel, > > Works for me see below. > > exten => _1NXXNXX,1,SetCIDNum(55,a) > exten => _1NXXNXX,2,Dial(IAX2/USER:[EMAIL PROTECTED]/${EXTEN},30,Tt) > > Kevin > > -Origina

Re: [Asterisk-Users] Asterisk With Broadvoice

Great! It works now!! Thanks so much. Roger Hanson wrote: - Original Message - From: "Robert Webb" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" ; <[EMAIL PROTECTED]> Sent: Friday, February 25, 2005 2:49 PM Subject: Re: [Asterisk-Users] Asterisk With B

RE: [Asterisk-Users] SetCIDNum using SIP? Ignore my last post

Nabeel, Ignore my last post. Missed the SIP part of your question. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Collins Sent: Friday, February 25, 2005 4:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk

Re: [Asterisk-Users] Asterisk With Broadvoice

- Original Message - From: "Robert Webb" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" ; <[EMAIL PROTECTED]> Sent: Friday, February 25, 2005 2:49 PM Subject: Re: [Asterisk-Users] Asterisk With Broadvoice On Fri, 25 Feb 2005 14:42:09 + "[EMAIL PRO

RE: [Asterisk-Users] SetCIDNum using SIP?

Nabeel, Works for me see below. exten => _1NXXNXX,1,SetCIDNum(55,a) exten => _1NXXNXX,2,Dial(IAX2/USER:[EMAIL PROTECTED]/${EXTEN},30,Tt) Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Friday, February 25, 2

Re: [Asterisk-Users] Fedora Core 3?

Race Vanderdecken wrote: I am running on an Intel Pentium 3, 1.5 GHz, mother board stuck inside an old E-machine case and it is very happy... (I only wish I could find a Okidata B4250 printer driver or a PCL-6 I could understand.) http://www.linuxprinting.org/pipermail/okidata-list/2004q2/000359

Re: [Asterisk-Users] Asterisk With Broadvoice

On Fri, 25 Feb 2005 14:42:09 + "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote: OK, After checking into this, I have found the following: I can set it up so either incoming or outgoing sip calls on this trunk work but NOT both. The "sip show registry" command shows everything as it should be.

RE: [Asterisk-Users] Asterisk With Broadvoice

Here is a copy of my config that works great with broadvoice. I also have an AGI that I wrote to verify country codes so your users can't call countries that aren't included in broadvoices plan. If you want that too, just let me know. Sip.conf ---

[Asterisk-Users] SetCIDNum using SIP?

I am experimenting with my * server to use SIP with my long-distance providers instead of IAX, so that the media path is from the end user straight to the provider's gateway (hopefully reducing my bandwidth consumption). I have it working with VoicePulse Connect but SetCIDNum doesn't appear to work

Re: [Asterisk-Users] Asterisk With Broadvoice

OK, After checking into this, I have found the following: I can set it up so either incoming or outgoing sip calls on this trunk work but NOT both. The "sip show registry" command shows everything as it should be. The section from my sip.conf is as follows: [Broadvoice] username = 2x ty

[Asterisk-Users] E100P to Valiant E1-PRI GSM gateway

Looking for zaptel/zapata configuration parameters to successfully communicate with a Valiant GSM gateway as above. Surely someone has done this? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 02/22/2005 ___

Re: [Asterisk-Users] Transposed ringing

On Fri, 25 Feb 2005 12:09:16 -0800 Trevor Peirce <[EMAIL PROTECTED]> wrote: I don't suppose anyone might know why I hear ringing transposed over itself when I place a call out via PRI? SIP to SIP is fine SIP to IAX is fine SIP to PRI is always transposed I mean sometimes you don't notice it much

RE: [Asterisk-Users] Festival - Asterisk@home

Title: Festival - [EMAIL PROTECTED] figured it out.  Stopped using the example festival-weather.script.pl and used the festival-script.pl that is in the directory already.   Works good.  Is the voice customizable?  Does memory on the box play a part in quality?   Thanks, Wiely   From: [EMAI

Re: [Asterisk-Users] WebVMail Woirks but No Audio

Hi Assaf - Already did that - the audio app location shows as a broken link on the page and plays nothing. On Fri, 25 Feb 2005 14:06:41 -0500 "Assaf Benharoosh" <[EMAIL PROTECTED]> wrote: > I had this issue- it's security on the files. I put a cron job that do > /bin/chmod 777 /var/spool/as

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 7, Issue 304

Daniel Nystrom wrote It seems like the Radio discussions is closing in on something I was interested in. How about controlling 30 2-way radios via E1 and 30-channel "Mux" (channel bank?) with E&M signalling? I think the Mux uses CAS and each channel has Audio out, PTT, Audio IN, Busy. 6-wire connec

[Asterisk-Users] Transposed ringing

I don't suppose anyone might know why I hear ringing transposed over itself when I place a call out via PRI? SIP to SIP is fine SIP to IAX is fine SIP to PRI is always transposed I mean sometimes you don't notice it much because it's lined up right, but other times you'll hear a really long ring

Re: [Asterisk-Users] Festival - Asterisk@home

Title: Festival - [EMAIL PROTECTED] Does your Festival installation work ok?  (run the tests/example scripts that came with the installation).  I installed Festival, and according to the installation scripts all went well, however none of the tests/example routines would work - I kept getting

Re: [Asterisk-Users] How does the g.729 registration program work?

On Fri, Feb 25, 2005 at 11:24:21AM -0600, Steven Critchfield wrote: > It is based on a machine unique key created by querying your hardware. > You will not be able to share your licenses between machines. You will > need to buy licenses for each machine you deploy on. You misunderstand. Ofcourse I

[Asterisk-Users] Festival - Asterisk@home

Title: Festival - [EMAIL PROTECTED] Hello All, I installed [EMAIL PROTECTED] with no problems whatsoever.  All features so far work great. However, I have been trying to setup the festivval weather AGI script and it won't work. I see the script fire off in the CLI and it completes with

[Asterisk-Users] Video Support Not Working

Title: Video Support Not Working Hello, I have a couple of video phones that I am trying to get setup.  I have used these phones with sipphone.com and they work great.  Now I am trying to get them to work with my * server and I am having problems.  The voice portion seems to work fine, but

[Asterisk-Users] Wheres the Math application

From the docs on this command it should be available in 1.0.1. I don't find it under the apps directory even after doing an update -d (which I understand will add missing files or diretories) I have also downloaded ver 1.0.5 and looked in its apps directory. It isn't there either. Any suggest

RE: [Asterisk-Users] WebVMail Woirks but No Audio

I had this issue- it's security on the files. I put a cron job that do /bin/chmod 777 /var/spool/asterisk/voicemail/default -R evey 1 minute, but there may be a cleaner solution. Assaf Benharoosh MCP, MCSA, MCSE [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EM

[Asterisk-Users] HELP NEEDED ASTERISK AND MEDIATRIX 1102

Hello all, Hi I would like to know how to configure a Mediatrix 1102 box to work with my asterisk box. I have analog phones that i would like to connect to my Mediatrix box and then connect the Mediatrix box to my asterisk box. My main problems come from the fact that I have limited experience wit

Re: [Asterisk-Users] FW: Getting PHP Config to work?

On Fri, Feb 25, 2005 at 04:43:50PM -, C. Tomlinson wrote: > Hi, > > Thanks for the batchfile type, it's a handy one. > > I'm not editing over the internet, just local LAN for testing ATM. Protected > via firewall. > > Would it not be fairly secure using a combination of the following: > .hta

Re: [Asterisk-Users] How does the g.729 registration program work?

On Fri, 2005-02-25 at 09:25 -0800, Mik Cheez wrote: > There is an open source version of the license: > > > You can view the licensing information at the following: > > > more details can

RE: [Asterisk-Users] Fax on Asterisk

Perfect. Thanks! Found lots for incoming and that filled the gap for out going. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, February 25, 2005 11:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Sub

Re: [Asterisk-Users] Fax on Asterisk

On February 25, 2005 12:57 pm, Wiley Siler wrote: > Is it possible to send an email to Asterisk and have it parse the email > or an attachment and send it out as fax? Open up your web browser, go to www.google.com and enter "asterisk send fax". That will get you info on how to send a .tiff file

Re: [Asterisk-Users] "click to dial extension number" functionality ?

By any web-user (ms explorer) to be able to call from a web-page to a certain number/extension connected to one specific asterisk. maybe this php script help you (switch caller/called and modify Exten:) --originate.php-- CALLER CALLED $socket = fsockopen($astip,"5038", $errno, $errstr); f

[Asterisk-Users] Fax on Asterisk

Title: Fax on Asterisk Is it possible to send an email to Asterisk and have it parse the email or an attachment and send it out as fax? Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailm

RE: [Asterisk-Users] VoIP/Asterisk presentation

Anyone able to get to these I am unable to get to them. -Original Message- From: Duane [mailto:[EMAIL PROTECTED] Sent: Friday, February 25, 2005 10:14 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] VoIP/Asterisk present

Re: [Asterisk-Users] Directory config...

> How do I config Asterisk so when the directory cmd is used, the name of > the found entry comes from a pre-record gsm file instead of being spelled > letter by letter? If the user as recorded is name, this file will be used. When it's not recorded, * will spell it. Dial to your voicemail and nav

Re: [Asterisk-Users] How does the g.729 registration program work?

There is an open source version of the license: You can view the licensing information at the following: more details can be found on http://www.voip-info.org Steven Critchfield wrote: On F

Re: [Asterisk-Users] How does the g.729 registration program work?

On Fri, 2005-02-25 at 16:59 +0100, Martijn van Oosterhout wrote: > I'm asking because I'm planning to install multiple machines from the > same image and I need to know what file(s) I need to backup/restore to > make sure I don't lose my licences in the process. The only options I > can think of ar

[Asterisk-Users] Speex transcoding for Cisco / Polycom

Hi guys, I have a weird problem, and I have encountered a few other people with the same issue. The problem is this: Whenever I make a call from my IAXy (g711ulaw) to my server, and then my server transcodes to speex and sends it to a remote asterisk server, audio is perfectly fine. The same

Re: [Asterisk-Users] cascaded ringing

On Fri, 2005-02-25 at 12:36 +, Julian J. M. wrote: > You could add > exten => 1,2,Goto(context,2,2) > > But I don't know what will happen when, after 5 secs, dial SIP/2 is > executed again... I think you are on the right track. If SIP considers ringing as busy then you can cascade through yo

RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI

You 're right, there are some security issues using using sudoers and system commands. If the asterisk server is reachable from the outside over http or other unsecure protocols, it would be really dangerous. But in a trusty intranet-environment, where firewalls block every attempt to access the a

Re: [Asterisk-Users] "click to dial extension number" functionality ?

El 25/02/2005, a las 12:10, Terje Myhre escribió: By any web-user (ms explorer) to be able to call from a web-page to a certain number/extension connected to one specific asterisk.   Almost as a web-based “auto-attendant” functionality.    Hence: 1. surf to the specific web-site

[Asterisk-Users] Avaya Partner ACS3 and Asterisk

Im sorry if this has been asked before as I couldnt seem to find it. I have an avaya partner ACS r3 system that I want to be able to hook asterisk into with a x100p card, into and use asterisk to tie into a voip provider then be able to dial (or connect) to an extension like an intercom function a

Re: [Asterisk-Users] T.38 fax summary

Mark, In the time it took to write all that you could probably have read up enough about T.38 to realise you were talking complete rubbish :-) Regards, Steve Mark Eissler wrote: On Feb 25, 2005, at 10:20 AM, Lee Howard wrote: In a traditional analog fax you have modulated audio data, that is, th

[Asterisk-Users] Avaya Partner ACS3 and Asterisk

Im sorry if this has been asked before as I couldnt seem to find it. I have an avaya partner ACS r3 system that I want to be able to hook asterisk into with a x100p card, into and use asterisk to tie into a voip provider then be able to dial (or connect) to an extension like an intercom function a

RE: [Asterisk-Users] FW: Getting PHP Config to work?

Hi, Thanks for the batchfile type, it's a handy one. I'm not editing over the internet, just local LAN for testing ATM. Protected via firewall. Would it not be fairly secure using a combination of the following: .htaccess file VPN? https access? Limit apache to only allow certain IP's? And the p

RE: [Asterisk-Users] FW: Getting PHP Config to work?

Thanks for the chmod, it definitely needed that! I didn't have to change the etc.sudoers file though. I'm running Debian, via the great Xorcom rapid installation. I didn't change the permit lines either as this is just attesting box and im not worried about security. -Original Message- F

Re: [Asterisk-Users] HDLC Bad FCS / HDLC Abort

tim panton wrote: [..] Good luck with it. I think I am lucky! ;-) We resolved the problem changing the Mother board to one with an Intel chipset. The first one had Via chipset. 1) At first, we changed the mother board by another with Intel chipset. This one can set udma2 (or udma3) from BIOS. It is

RE: [Asterisk-Users] FW: Getting PHP Config to work?

Thanks to all your help I have now got it working great. I have written a quick howto which I plan to add to the wiki if people approve? Take a look at: http://www.burntwires.com/asterisk/Install%20PHP%20Config.htm (Please excuse the bloated html) Please leave any feedback and then I will a

[Asterisk-Users] WebVMail Woirks but No Audio

Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET /vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default &password=1

Re: [Asterisk-Users] T.38 fax summary

On Feb 25, 2005, at 10:20 AM, Lee Howard wrote: In a traditional analog fax you have modulated audio data, that is, the data stream is converted into an audio representation by the transmitter, and the receiver demodulates the audio stream to produce the data stream. A lot of data gets packed i

[Asterisk-Users] Directory config...

Hi all, How do I config Asterisk so when the directory cmd is used, the name of the found entry comes from a pre-record gsm file instead of being spelled letter by letter? Regards, Francois Random Thought: --- All of us failed to match our dreams of perfection. So I rate us on the

[Asterisk-Users] How does the g.729 registration program work?

I'm asking because I'm planning to install multiple machines from the same image and I need to know what file(s) I need to backup/restore to make sure I don't lose my licences in the process. The only options I can think of are: - There's a config file, though I've seen no mention of it - The actu

RE: [Asterisk-Users] CDR writing incorrect data to pgsql tables

> For MySQL and other glorified flat-file databases, you would > need to postprocess the data. You may feel more confident > skipping triggers and doing this anyway. > > > So by that any calls that go out over the net using IAX to > the telco > > are considered digital and will report correctl

Re: [Asterisk-Users] FW: Getting PHP Config to work?

On Fri, Feb 25, 2005 at 01:52:21PM -, C. Tomlinson wrote: > Richard, > > I have been using WinSCP to transfer files across easily without messing > with FTP accounts. I had not found that feature, many thanks for pointing it > out :-D > > I will definitely use this from now on until I find a

Re: [Asterisk-Users] FW: Getting PHP Config to work?

On Fri, Feb 25, 2005 at 03:15:34PM +0100, Michiel van Baak wrote: > On 15:04, Fri 25 Feb 05, Eivind Trondsen wrote: > > Richard Folwell wrote: > > > > > >Look at WinSCP: > > > > > > > > >It is (almost) worth installing Windows just to be able to use it. :-) > > >If anyone knows of anything simi

RE: [Asterisk-Users] Fedora Core 3?

I am developing voicemail and SIP and RAIDUS code for Asterisk Code on the Fedora Core 3 and having no problems. I am running on an Intel Pentium 3, 1.5 GHz, mother board stuck inside an old E-machine case and it is very happy... (I only wish I could find a Okidata B4250 printer driver or a PCL-6

RE: [Asterisk-Users] Asterisk and 723,729

The Cheapest way is to purchase 2 licenses, or in multiples of 2 If you need more, from Digium.   You will be beating a dead horse and a dead carriage and a dead driver if you try to get around G729 licensing. You only need a license for each answer and originate session that uses g.729

RE: [Asterisk-Users] FW: Getting PHP Config to work?

Hi, I'm not sure the way to change it, but when I d/l it from http://asterisk.espia-net.net/horde/chora/cvs.php/phpconfig/cls_phpconfig.ph p?login=2&asterisksess=5c8e63576772790cfc2e1dbce354e04d I had read about the problem with fget's, but presumed this change was the correct one. However it l

Re: [Asterisk-Users] Re: FRS and GMRS via *

There are GMRS radios that support frequency splits... I dont think FRS does. -- Mike - Original Message - From: "TC" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, February 25, 2005 9:10 AM Subject: Re: [Asterisk-Users] Re: FRS and GMRS

Re: [Asterisk-Users] CDR writing incorrect data to pgsql tables

James Bean <[EMAIL PROTECTED]> wrote: [...] > Is the kludge done at the software side when the data is pulled out > for accounting and being under say 45 seconds is a no answer or > busy? Or is there a tweak that can be done at the database itself? Since you're using PostgreSQL, you can use a trig

RE: [Asterisk-Users] Vonage <---> Asterisk Complete Config

Must have missed a few messages :) Vonage always allowed this on "softphone" lines. Those are $10/month with metered usage (100 min included). They also require a "hardline" (ATA) as the primary line on the account. It's a working crutch for those folks who need a DID in a rate-center only vona

Re: [Asterisk-Users] ATA that actually work with T.38

On Feb 25, 2005, at 7:55 AM, Steve Underwood wrote: If you understand what T.38 is you will understand which problems it addresses (summary: it is important for solving some problems, but nothing solves them all). Most people who post about T.38 don't actually have much of a clue about it. I thi

RE: [Asterisk-Users] ATA that actually work with T.38

[EMAIL PROTECTED] wrote: > For T.38 passthrough between RTP channels it doesn't need to know a > great deal. There are some pitfalls, though, due to dumbness > in the T.38 > spec. > > Are you actually working on this? Yes, well, with a lot of other things, so progress is erratic. I've got to so

Re: [Asterisk-Users] Working SIP phone for linux and windows

> I have yet to discover a software package that would both register and > have ulaw codec. The SIP communicator (Java) came closest to usable, > but didn't have the ulaw codec working. What do you use for > communications? for SIP you can use X-Lite : http://www.xten.com/index.php?menu=products&s

RE: [Asterisk-Users] SIP Errors

Hmmm, Looking directly at the .../channels/chan_sip.c code does not get any clues. Switch( resp ) ... ... case 480: /* Temporarily Unavailable */ case 404: /* Not Found */ case 410: /* Gone */ case 400: /* Bad Request */ case 500: /* Server error */ case 503: /

Re: [Asterisk-Users] HELP NEEDED! - Asterisk GUI

Hi On Thu, Feb 24, 2005 at 11:41:41AM +0100, Hecken, Guido wrote: > >Secondly, is the statement no.2 a line a need to change in a given file? > You have to change/verify some settings in phpconfig_init.php . > Look for fakeuser=admin. > Set $reset_cmd = "./asterisk.reload"; > Be shure, the script

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