similar to: X100P FXO PCI Card + Incoming Fax

With the following program: #!/bin/sh # mailfax: program to email received fax as pdf FAXFILE=$1 RECIPIENT=$2 FAXSENDER=$3 FAXID=`basename $1|cut -d "." -f1,2`.pdf FAXTXT=`basename $1|cut -d "." -f1,2`.txt tiff2pdf $FAXFILE > $FAXID sendfax.pl $FAXID $RECIPIENT $FAXSENDER $FAXFILE #end of program If I execute the following from the command line: mailfax

Sip to sip calls are fine, both local on Asterisk and over a SIP gateway, however some people who call on the PSTN line say we are very queit and vice versa, can the volume be turned up on the PSTN line? The volume buttons on the VoIP phones only turns up the others voice, so this is a fix for us, but how do I make our voices louder for the people on the PSTN line? Thanks. Paul.

Hi, I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there are 3 BT lines connected directly to these ports. One of the lines has BT FeatureLine Compact and this is the line I am having problems with, the other 2 lines are working perfectly, detecting CID, answering incoming calls and placing external calls via SIP devices. I am receiving a error log entry: chan_zap.c:

Hi all, I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and 1XFS modules. The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM. Sangoma A200 has 3 analogue PSTN lines connected. This server is based in Office 1, with 5 users all with a Linksys SPA942 VoIP Handset. There is another Office (Office 2) connected to here using VPN. There are two users in Office 2 with the

We recently received a bug report about a compiler issue when using GNUTLS: http://bugs.gentoo.org/show_bug.cgi?id=67628. I can confirm the same bug using 0.99.11. Help? Thanks, g2boojum -- Grant Goodyear Gentoo Developer g2boojum at gentoo.org http://www.gentoo.org/~g2boojum GPG Fingerprint: D706 9802 1663 DEF5 81B0 9573 A6DC 7152 E0F6 5B76 -------------- next part -------------- A non-text

Robert, > Is there a product to simulate a PRI trunk? (Robert > Goodyear) TDMoE emulates a T1. ;) Once the TDMoE link is up, Asterisk just sees 24-lines that appear to be a T1 instead of having to deal with all of the complexities of VoIP. This is useful, since probably 75% of the utility of VoIP is really just the fact that it can run over a network. It's also handy because it

Does anyone know if there's a service out there to -- for a fee -- inject our DID into the LEC's CLI database so a called party gets our associated name? /rg

Does anyone know of a way to simulate the signaling of a PRI trunk for testing/setup purposes? I realize this may be a rather naive question, but I was wondering if you could take a TE110, for example, and using a crossover cable (or not?) and some means of emulating the NI2 signaling protocol connect it to another TE110 on another machine to test and verify an installation before the telco

Robert, Essentually I want to be able to have Server B dial the extensions connected to server A as well as route calls to the outbound route on Server A. Server B will have little to no knowledge of what is on Server A. I just want it to dump the calls off. For some reason I keep thinking this was a PRI type of thing. Like there was a module that loaded up as a fake PRI that your

I am currently caching a page that indexes blog entries. Paginate was used to break the entries up. Now when you click on next at the bottom of the page it is only reloading the first page. Im hoping, against odds, that someone here knows of a way to make paginate and cache play together. -- Posted via http://www.ruby-forum.com/.

I know a SIP client behind a NAT trying to peer with Asterisk behind another NAT is troublesome. Has anyone had any luck doing this by interfacing Asterisk to the WAN using 1:1 NAT translation to give it a public IP while still firewalled? In my instance I'm using m0n0wall, but this is a hardware-neutral question. Thanks. -- Robert Goodyear Brand Up LLC http://www.brand-up.com

I am trying to use caches_action and the agile book says that it is keyed off the URL, however it does not seem to pick up the URL parameters. http://localhost:3000/controller/action/id?foo=bar and http://localhost:3000/controller/action/id?foo=foobar Returns the same page. Anyone looked into how to add URL parameters to the cached key? Also, where do those file get stored? Thanks Dave

Hi, Just playing with a couple of Budgetone 102 phones and they are pretty good for the price. The only problem i'm having at the moment is when I get a voicemail on the Asterisk box the LCD flashes. Dialing *98 goes to the VoiceMail Manager, and asks for mailbox, I enter 201, then asks for password, enter my voicemail password set in the Extensions -> webadmin, then hit the

I'm all up for reading and looking round for people in the same boat to try and solve the issue together, but there appears to not be large community yet, just the asterisk mail lists. I got Asterisk working with X-Lite great now for internal calls and also calling land line numbers etc. The two problems i'm currently having are: 1. When someone calls in the phones ring 3 times then

(Sorry, but my english is not good) Hi, I have a problem with my new asterisk instalation. I search in google but I couldn't find nothing. Here's the thing. Before, we have 2 asterisk servers, each one with a E1 card. one with a Digium TE105 and the another with a A104 and we have a very simple setup. A Linux IBM X4300 Server is running CentOS 5.4 + Asterisk 1.6.2 (one month ago I

I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all. Just as a quote note, can I thank everyone on this list. I find my self finding pretty much every answer I am looking for on here. And a big thanks to all thoughs helping us out. Mass Respect :) Ok, I'm using a SIP provider (SipGate UK) to do my international dialing etc, working great from extension 8 on phones. However some more friends/contacts have started using SipGate also, and

I also want to try that asterisk guide. But i'm not sure if i understood it correctly. What exactly do i need to do? Do i need to compile Asterisk with the spanDSP plugin or just configure extensions.conf? The URL to spanDSP in the guide wasn't working. I also use a traditional internet line to recieve calls and hopefully i will get Fax working soon. This is so confusing. Thanks, Arne

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- > inject our DID into the

Hi all, Does Asterisk support multi thread? I mean: Is it possible to do one of the 2 following scenarios: 1. Play a low background music when the user record his/her voice 2. If the first scenario is not possible, can we play two music stream at the same time? i.e: using MP3Player to play a music file and at the same time play the recorded voice of the user. Thanks in advance for any