VOIP

fwd numbers

me 275052
steve 754819
dean 257532

• Asterisk conference http://Astricon.net/
• Internet VOIP conference http://www.tmcnet.com/voip/conference/

I currently use Packet8 service at home, have for a couple years now. It works fine but I am now diving in deeper.

I am also building a system for a small business, which will probably use Linksys equipment.

I am playing with softphones now.

pulver communicator (from fwd site) (Windows)

ekiga (for Linux) could not get it going because the linux box has no mic input! (A Shuttle SB95)

Adore softphone (Windows)

Hardware

PSTN gateways

Linksys SPA-400

This seemed like it might be an interesting device because it's open source. It is a pain to work with, I have given up on it. It could be made into a really nice product if it had decent software on it. It probably works fine as intended, in an all Linksys system. You can download the source tarball from the Linksys site.

version 1.0.0.2

It looks like it's based on Monta Vista Hardhat Linux

• MIPS processor
• ?? Flash
• ?? RAM
• Linux 2.4 kernel
• Busybox
• Asterisk PBX software
• thttpd
• /usr/sbin/lightbox controls LED's
• dropbear ssh

Digium X100P

These are fine if you only need one line, cheap on ebay.

Digium TDM400

This is the card I am working with now.

Sipura SPA3000

I'd like to try one of these as a single line solution.

Asterisk

You might want to use an IDE/CF adapter that can hold two CF cards. This way you could stick in a smaller second card -say- 32 or 64MB and mount stuff like /etc/asterisk, /var/log, /var/run and voicemail etc on that one, possibly leaving the main card mounted read-only for most of the time.

http://www.limeylinux.org/

The Linksys SPA-400 uses flash on board for the operating system and then mounts /var on a USB stick. It sticks out way too far but that's another story.

Using Digium cards for FXS/FXO

Zaptel modules

Use the command genzaptelconf -d to generate new config files. Generates /etc/zaptel.conf and /etc/asterisk/zapata-channels.conf files.

Echo cancellation and gain settings on FXO lines

This page covers both of these topics:http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html

This one is interesting too: http://www.rowetel.com/blog/?p=18

1. Edit /etc/zaptel.conf to adjust receive and xmit gain levels.
2. Use ztmonitor channel to see levels during a call.
3. Use asterisk -r and the reload command to change settings during a call.

Links

Asterisk

Free World Dialup