Packet8 -BPG510

Hello all.

I’ve been a linux user since 1997 and have been toying with the idea of setting up an asterisk box for quite a few years. I’ve finally decided to go for it, but have a few questions first.

I currently have VOIP from packet8. My VOIP device is a BPG510.

Question #1 - Will my device work?
Question #2 - What digum card will I need? Are there different ones for analog lines vs. VOIP lines?

Thanks guys/girls!

as I recall packet8 is ‘locked’ which means that you can only use packet8 with your adapter. For * this is very sub optimal but it will work.

OTOH, if you have a different provider like viatalk quantumvoice or broadvoice, you can do ‘BYOD’ (bring your own device) which means they just give you the SIP password and * connects directly via VoIP. You should email packet8 and ask for this, don’t expect to get it but it’s always good to be heard :smile:

To use your existing service will be no different than an analog phone line. You will need an FXO port, and run a phone cable from the FXO port to your packet8 ATA (analog telephony adapter, the voice device).

  1. you will probably want one of the digium tdm400 series. The TDM400 is a modular 4-port analog card, each port can be enabled as FXO (connects to a phone LINE, red module) or FXS (connects to one or more phones and provides dialtone, green module) by adding the appropriate modules. The TDM400 itself is available pre-configured in every possible configuration, you would probably want the one FXO config. If you buy a pre-configured TDM400 you can purchase the modules later to bring additional ports online in whatever format you need down the road.

Asterisk does not require an interface board for VoIP, VoIP goes over standard Ethernet and uses your machine’s standard 10/100(/1000) network adapter. In a pure-VoIP setup, the network adapter is all that is required. Interface devices are only required for connecting to TDM-based telephone systems like analog (POTS) or PRI/T1.

Thus, it is ideal if your provider will give you the SIP login info for your account. This can be simply plugged into * and you are good to go. With a locked provider however, you are required to use the ATA and an analog phone cable. This results in a slight loss of quality, but the main problem is that Asterisk doesn’t entirely deal with certain analog services like call waiting. It’s possible to make it work but not nearly as well- with a BYOD provider, the second call can come in and ring other phones, and you can have two calls going at once. Not so with an ATA- your one analog cable means one call only.

If you wish to experiment with VoIP, the above mentioned providers usually have a BYOD trial plan where you pay a smaller fee of around $5-10/mo with no contract to see if you like it, this gets you around 500min outbound and free incoming.

Hope that helps!

Packet* does not “work” with asterisk, expect in using the ATA as IronHelix has stated, They do have a software client but they charge for the calls made using it, kind of dumb but what can you say.

I have used my Packet8 ATA as IronHelix has laid out, it worked fine with a bout week of tweaking. I had the ATA feeding three phones just fine but I had to run a new homerun line to get the card to hang up (the ole lines had about three cuts and patch jobs in them)

All in all it worked fine, I am dropping Packet8 as of this month, as the low amount of outbound calls I make using the 8x8 service; it is cheaper using Voicepulse than paying the $ 25.00 a month for the unlimited, as most of my calls are inbound on that number and switched in to a unlimited inbound did for 11.00 a month.