VOIP
fwd numbers
me 275052 steve 754819 dean 257532 agi 823170
- 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
I tried out a Linksys SPA-400 with 3 PSTN analog phone lines and an Asterisk server running on a recycled PC. It probably works fine as intended, in an all Linksys system but I was never able to get it to work reliably with Asterisk. The irony is that the SPA-400 runs Linux and Asterisk internally.
You can download the source tarball from the Linksys site.
version 1.0.0.2
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.
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.
Setting up
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
- Edit /etc/zaptel.conf to adjust receive and xmit gain levels.
- Use ztmonitor channel to see levels during a call.
- Use asterisk -r and the reload command to change settings during a call.
Web interface
Voicemail
User interface
On delete of message, move it to Trash/
Administration interface
- Should trashed messages be deleted or archived? How often?
- Should messages marked as old be deleted or archived? How often?
- Per user control of Trash/ ?
Links
Digital signal processing and Spandsp: http://www.soft-switch.org/
The Blackfin project http://rowetel.com/ucasterisk/
