Tech Friday is our occasional Friday afternoon dweeb-fest, where if we are going to publish actual programming code (rarely), or get technical, it tends to happen on Friday.

Since installing Trixbox, the pre-configured version of Asterisk I’ve been playing with all the features. I’ve got it connected to a single phone line, with a simple automated attendant, “press 1 for Larry, press 2 for Bob” kind of thing and that part seems to work fine. I’ve got two Grandstream Budgettone 101 phones working fine. They have message ready lights, tell the date and time and will show the caller-id on their LCD read-out.

I read the O’Reilly book about Asterisk, which serves as the Asterisk user manual, and it was very helpful in figuring out the intricacies of trunks, routes, and extensions, and how these all fit into a dial plan. But now I’ve hit the wall, and am trying to figure out two show-stoppers:

Hanging up the line

Sometimes the hardware phone line doesn’t “hang up” after a call. Example: Somebody calls through the landline, leaves a voice mail and then hangs up. There are times when the hardware card doesn’t hangup. Searching through through the archives, I found several mentions of this, so far with no good solutions. It applies to both the X100T Digium cards, and apparently the T4xxxx cards as well. Supposedly the Sagamore cards are more reliable. But, I’m hesitant to shell out another couple hundred bucks for more cards.

Configuring software “trunks”

As an alternative to hardware trunks, (destinations for calls placed outside your own organization), you can configure either free or (usually) paid-for internet destinations where your call is sent and then connected to other subscribers or to landliness. Ultimately, of course, this is indeed what you want to do…any phone system is useless if you can’t connect to other phones. I first tried Free World Dial-Up a free service that was one of the first available termination services. My box seemed to register with this without problems, but any calls sent out to the destination were unanswered.

So then, I subscribed to VoicePulse using their plan for Asterisk. VP will provision your Asterisk service with four trunks, two for the IAX2 protocol, native to Asterisk, and two SIP trunks. The SIP trunks appear to register, but the IAX2 trunks do not. According to the VoicePulse tech support, I may have problems with closed ports at my router.

You need to have port 5060 open for SIP and/or 4569 UDP open to use IAX2. Using my D-Link DI-604 router I tried several different configurations; I put the Asterisk machine in the DMZ, which means that is should be exposed to anything coming to the router, and I forwarded specific ports to the Asterisk box. No cigar.

I then checked the availability of the ports using Steve Gibson’s Shields-Up. These show both ports as “closed”. Uh oh. It may be related to the router or possibly something else. However, if they have been closed by Comcast (our new owner who has taken over from Adelphia Cable) then I may be SOL.