http://blog.voipsupply.com/asterisk-hardware/iax-phone-contest

Customers of Geek would know that now a big push for Geek is PBX, particularly the VOIP PBX, we offer 3 brands and I will get into more detail about that later.

One brand we offer to the very small business is Asterisk. This solution, like any VOIP sultion is 90% networking and 10% telephony.

Well one of the things that makes the networking side of it so hard is that SIP, Skinny and other phone control protocols are cumbersome and messy.

SIP (Session Initiation Protocol) uses a shotgun approach, via RTP (Real Time Protocol) to get the message across, in Astersisk the default ports that need to be punched through your firewall are: 10,000-11,000 (look in your /etc/asterisk/rtp.conf)

Most routers and firewalls don’t like this and can lose track easily. Particularly when NAT (network address translation is involved)

The answer to this with SIP phones is to:
a) limit your self to one SIP phone per NAT(ted) end point (ie one phone per remote site)
or
b) use IPSEC tunnelling to provide a “routed environment”

While IPSEC is more secure, that lax security only applies to internal (extension to extension) conversations. External conversations are never secure.

Now there is another option!

http://blog.voipsupply.com/asterisk-hardware/iax-phone-contest

Voipsupply in the US has released a new IAX2 compatable phone, and while these have been available in the past they have been of the cheap rubbishy “yum cha” variety. The new phone seeks to change all that.

IAX stands for Internetwork Asterisk eXchange and it is what Geek uses to connect multiple Asterisk servers together (when you need connectivity between two sites). Also one of our Voice carriers support this protocol and via it, we can easily deliver 100 number blocks cheaply and with a minimum of configuration.

Getting back to the point, with this new phone we can have a phone that can easily be taken anywhere and plugged in to a broadband connection, with NO configuration required AT ALL on the firewall, in almost all cases you can start talking instantly. And you can have as many phones as you like on the one connection without the need for a remote server or IPSEC tunneling!

I will watch this phone closely and asses the DSP (digital sound processor) quality and it’s speaker phone. If the sound stacks up, it could be a useful addition to the Geek PBX offering!

Installing wideband (g722) codec translation in Asterisk 1.4.x

To install g722 as an available codec first install the backport patch: (usual disclaimers apply – it worked for me and did not ruin my pbx, use at your own risk)

cd /usr/src/asterisk
wget http://carlton.oriley.net/drupal/fil…7.1-g722.patch
patch -p0 < asterisk-1.4.7.1-g722.patch
make
make install
amportal restart

to see if it worked, jump into the console:
Asterisk -r
and then type
core show translation

along the bottom of the table alongside g722 you should see a row of numbers (the lower the better ) if you see dashes, the install didn’t work.

You will of course need to use the allow=g722 statement for the protocols you require:
sccp.conf, sip_custom.conf & iax_custom.conf
Plus any extensions for which you have used the disallow statement.

Bear in mind of course, there aren’t too many phones that support wideband as yet…