VOIP: Difference between revisions

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I'd like to try one of these as a single line solution.
I'd like to try one of these as a single line solution.
==== Sipura SPA3102====


I have a SPA3102 now that I picked up for the [[Chintimini Wireless Project]].
I have a SPA3102 now that I picked up for the [[Chintimini Wireless Project]].
=== Handsets ===
Ones that I have tried include
Grandstream BT-100 -- cheapie
Linksys/Sipura SPA-841
Linksys SPA-941
Polycom IP-550
Polycom IP-650
What has paging / auto-answer / intercom support??? [http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom]
Some phones support multi-cast. incl Aastra, Snom, Linksys
Aastra -- The ALERT_INFO variable works for 480i, 480i CT, 9133i, 9112i firmware 1.2.x or later [http://www.voip-info.org/wiki/view/Sayson+IP+Phone+Auto+Answer]
The bugs listed for this phone seem to preclude its desirability.
It's been replaced by the 9480i which is TOO EXPENSIVE
Grandstream GXP-2000 -- supports POE
Linksys SPA-841
Snom 300 supports POE and auto answer and multicast paging
http://www.snom.com/en/products/ip-phones/snom-300-ip-phone/


== Software ==
== Software ==

Revision as of 17:27, 7 January 2010

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

After a few months of wrestling with it, we bought a TDM-400 and sold the SPA-400 on eBay.

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. It's great.

Sipura SPA3000

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

Sipura SPA3102

I have a SPA3102 now that I picked up for the Chintimini Wireless Project.

Handsets

Ones that I have tried include

Grandstream BT-100 -- cheapie

Linksys/Sipura SPA-841

Linksys SPA-941

Polycom IP-550

Polycom IP-650

What has paging / auto-answer / intercom support??? [1]

Some phones support multi-cast. incl Aastra, Snom, Linksys

Aastra -- The ALERT_INFO variable works for 480i, 480i CT, 9133i, 9112i firmware 1.2.x or later [2] The bugs listed for this phone seem to preclude its desirability. It's been replaced by the 9480i which is TOO EXPENSIVE

Grandstream GXP-2000 -- supports POE

Linksys SPA-841

Snom 300 supports POE and auto answer and multicast paging http://www.snom.com/en/products/ip-phones/snom-300-ip-phone/

Software

Asterisk -- works well but pretty complicated for small installations. I have some set up notes here Asterisk.

Freeswitch -- http://www.freeswitch.org/

I am testing a set up on 8track

Kirk Bailey is experimenting with pfSense (http://pfSense.com/) which is a FreeBSD based firewall project. It is intended for embedded systems such as the Soekris. pfSense has a package management system and there is now a FreeSwitch package. I am going for a file server based solution right now so this is on hold for me.

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/

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

  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.

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/ ?

Service providers

Internet Telephony Service Providers

Links

Asterisk

Digital signal processing and Spandsp: http://www.soft-switch.org/

The Blackfin project http://rowetel.com/ucasterisk/