Small Office VOIP Setup

Voice Over IP, Asterisk, Handytone 488, Polycom Soundpoint 301

I'm a big fan of technology, open source, and low cost. I used to be more into telephones when I was a kid, too.

Asterisk has refreshed my interest in telephones. I took my first step into VOIP by purchasing the Handytone-488 ATA. It worked pretty well with Asterisk, but because of firewall problems with my Asterisk server, I ended up just connecting it to a SIP proxy service called Galaxy Voice.

I've been very happy with Galaxy Voice thus far. The one complaint I had was that I had to explicitly request to use g729 instead of g711 ulaw. Now that its setup like that, I'm very happy.

Next I setup an Asterisk server and relieved the 488 from active duty. This is what I use now. The asterisk server is behind a router, but it is setup as the DMZ host, and it updates the dynamic IP address when necessary. (Though I just signed up for a static IP, so that will soon be a thing of the past.)

Attached to the Asterisk server are two Polycom Soundpoint 301s. These phones are awesome. I'm going to use the Handytone 488 for home use from now on.

How do I manage faxes? Well I have an old standby telephone line for emergencies, and it also is my dedicated fax line. I don't have it setup for long distance though. For long distance faxes, I used to use 1010565, but now I use a calling card from PhoneHog. PhoneHog is a marketing website that gives you points for checking out advertisements. Kind of a pain, but not bad for some free minutes if you need to make a fax. It seems like some of the ads don't actually give you the minutes they promise though. The reporting is not too detailed so its hard to decipher.

I will also post my dial plan, sip.conf, iax.conf, and extensions.conf setup soon.

One problem my system has been experiencing is a dropped SIP registration for incoming calls. I think this has to do with a dynamic IP address and being behind a firewall. Therefore, I just ordered cable (much faster speeds anyway) which will allow me to put my Asterisk server directly on the internet, as well as on a static IP. I'll report back with how well this works.

Attempt this fix the Asterisk incoming call problem, or Asterisk incoming calls problem #1: I have changed the settings in sip.conf, specifically the defaultexpiry and maxexpiry. These settings refer to the registration expiration, and I guess that if the sip server I am connecting to has a stale registration, its not going to know where to send my incoming calls. I also set nat=no, because asterisk is now on the DMZ, and will in the future be right on the internet with no ip masquerading going on between asterisk and the internet.

For now, the settings are as follows:

maxexpirey=30 ; Max length of incoming registration we allow
defaultexpirey=15 ; Default length of incoming/outoing registration

These settings are much lower than the default. If they prove problematic, I'll increase them.

So far this seems to be working well.

The Soundpoint 301 is a rock solid phone, but it is kind of confusing. If you want to transfer a call, you have to press "More" -> "Trnsfr" -> "Send" -> "Trnsfr", which is way more complicated than it need to be. And to go into one-way speaker phone mode on an existing call, you need to press "Hold", then hang up, then "Resume Call".