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== Set up ==
== Wildsong's phone system overview ==


* A small Debian server ([[Bellman]]) runs Asterisk
* A small Supermicro server ([[Bellman]]) runs Asterisk on Debian
* Grandstream GXV3240 is the main office phone
* A Grandstream GXV3240 Android deskphone is the main office phone
* SPA2000 directs calls to a model 2500 in the kitchen and a generic analog phone in the studio, they share the same line.
* A Linksys SPA2000 directs calls to a Western Electric model 2500 in the kitchen and a generic analog phone in the studio, they share the same line.
* Elastic SIP Trunking service is provided by Twilio
* Elastic SIP Trunking service is provided by Twilio
* Inbound calls are directed to an extension that rings the SPA2000 phones, the GXV3240, and sends the call back out to my cellphone via Twilio.
* Inbound calls are directed to an extension that rings the SPA2000 phones, the GXV3240, and sends the call back out to my cellphone via Twilio.

Revision as of 00:30, 17 March 2017

Wildsong's phone system overview

  • A small Supermicro server (Bellman) runs Asterisk on Debian
  • A Grandstream GXV3240 Android deskphone is the main office phone
  • A Linksys SPA2000 directs calls to a Western Electric model 2500 in the kitchen and a generic analog phone in the studio, they share the same line.
  • Elastic SIP Trunking service is provided by Twilio
  • Inbound calls are directed to an extension that rings the SPA2000 phones, the GXV3240, and sends the call back out to my cellphone via Twilio.
  • SMS messages come to Twilio and gets forwarded to my cellphone via a TWMLBIN rule (Programmable SMS).
  • The Asterisk PBX handles voicemail.

We still have a GoogleVoice number that forwards calls to the Twilio number and to Julie's cellphone. I am relying on the Asterisk forwarding to get GoogleVoice calls on my cellphone.

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 GXV3240 (my favorite), GXV3275, DP715

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] The GXV324xx phones do, I have tested it and it works fine.

Some phones support multicast. incl Grandstream, 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

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/

Network

Setting up QOS on Mikrotik

Refer to http://wiki.mikrotik.com/wiki/Voip 

/ip firewall mangle
  1. First mark things as SIP
add action=mark-packet new-packet-mark=SIP_IN  chain=prerouting  in-interface=ether1-gateway  disabled=no passthrough=no \
    src-address=192.168.1.2 comment="Bellman"
add action=mark-packet new-packet-mark=SIP_OUT chain=postrouting out-interface=ether1-gateway disabled=no passthrough=no \
    dst-address=192.168.1.2 comment="Bellman"

add action=mark-packet new-packet-mark=SIP_IN  chain=prerouting  in-interface=ether1-gateway  disabled=no passthrough=no \
    src-address=192.168.1.43 comment="GXV3240"
add action=mark-packet new-packet-mark=SIP_OUT chain=postrouting out-interface=ether1-gateway disabled=no passthrough=no \
    dst-address=192.168.1.43 comment="GXV3240"

add action=mark-packet new-packet-mark=SIP_IN  chain=prerouting  in-interface=ether1-gateway  disabled=no passthrough=no \
    src-address=192.168.1.45 comment="SPA2000"
add action=mark-packet new-packet-mark=SIP_OUT chain=postrouting out-interface=ether1-gateway disabled=no passthrough=no \
    dst-address=192.168.1.45 comment="SPA2000"

add action=mark-packet new-packet-mark=ELSE_IN  chain=prerouting  in-interface=ether1-gateway  disabled=no passthrough=no \
    src-address=!192.168.1.2 comment="All Else"

add action=mark-packet new-packet-mark=ELSE_OUT chain=postrouting out-interface=ether1-gateway disabled=no passthrough=no \
    dst-address=!192.168.1.2 comment="All Else"

add action=mark-packet new-packet-mark=ELSE_IN  chain=prerouting  in-interface=ether1-gateway disabled=no passthrough=yes \
    src-address=192.168.1.2 src-port=80 protocol=tcp comment="Bellman Web Server" 

add action=mark-packet new-packet-mark=ELSE_OUT chain=postrouting out-interface=ether1-gateway disabled=no passthrough=yes \
    dst-address=192.168.1.2 dst-port=80 protocol=tcp comment="Bellman Web Server"

iptables -A twilio_whitelist -j ACCEPT -s 54.172.60.0/30 iptables -A twilio_whitelist -j ACCEPT -s 54.244.51.0/30 iptables -A twilio_whitelist -j ACCEPT -s 54.171.127.192 iptables -A twilio_whitelist -j ACCEPT -s 54.171.127.193 iptables -A twilio_whitelist -j ACCEPT -s 54.171.127.194 iptables -A twilio_whitelist -j ACCEPT -s 54.65.63.192 iptables -A twilio_whitelist -j ACCEPT -s 54.65.63.193 iptables -A twilio_whitelist -j ACCEPT -s 54.65.63.194 iptables -A twilio_whitelist -j ACCEPT -s 54.169.127.128 iptables -A twilio_whitelist -j ACCEPT -s 54.169.127.129 iptables -A twilio_whitelist -j ACCEPT -s 54.169.127.130 iptables -A twilio_whitelist -j ACCEPT -s 54.252.254.64 iptables -A twilio_whitelist -j ACCEPT -s 54.252.254.65 iptables -A twilio_whitelist -j ACCEPT -s 54.252.254.66 iptables -A twilio_whitelist -j ACCEPT -s 177.71.206.192 iptables -A twilio_whitelist -j ACCEPT -s 177.71.206.193 iptables -A twilio_whitelist -j ACCEPT -s 177.71.206.194 

/queue tree
add name="IN" parent=global-in priority=1
add name="OUT" parent=global-out priority=1
add name="SIP_IN" packet-mark=SIP_IN parent=IN priority=2
add name="SIP_OUT" packet-mark=SIP_OUT parent=OUT priority=2
add name="ALL_ELSE_IN" packet-mark=ELSE_IN parent=IN priority=8
add name="ALL_ELSE_OUT" packet-mark=ELSE_OUT parent=OUT priority=8

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/


Digium

Elastix

iPitomy

Qualmetrics

Trixbox

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