extending analog pbx port ?

Edwards Finance > Phones

Q. I'd like to extend an analog pbx station port from the office to my home. It should work like an off prem extension, that is when I go off hook at home, I should get pbx dial tone... when someone calls the extension, my analog phone at home should ring . I know there are a lot of boxes out there that do this, but they are hundreds if not thousands of bucks... I'm wondering if there's an inexpensive solution (as I'm paying for this on my own).

A. You can try a pair of Sipura SPA-3000's. One of the canonical examples they have in their FAQ's is how to set up the dialplan to do a "hotline". Eg. when you pick up one phone it automatically calls the some number. You can use that at the PBX-connected sipura to call your remote sipura whenever a call comes in. The remote sipura would be setup normally, with the pbx-connected one designated its outgoing sip gateway. For a bit of added simplicity I'd be sure to get two identical fxo/fxs units. You'll be screwing around with enough tricky settings in the units without having to worry about learning two totally different command sets. (There are probably other FXO/FXS units that can do the job too. I'm only familiar with the Sipura unit.) I have a SPA-3000 that I use to feed a POTS line into my asterisk (PBX). It works, but the reality of POTS lines is that it is impossible to feed them into a VOIP system without either causing a problem with reduced volume or if you crank the volume up, without introducing a bit of echo. If your pbx outputs a digital signal (PRI or BRI) this is probably the way you want to go. There are quite a few universities that have hooked their old PBX to either Cisco equipment or asterisk/SER and use SIP to connect things. In their case the goal is also to be able to connect to their PBX over the net, but only for the purpose of calling the users on their pbx. The setups should be similar enough that the configuration examples might prove useful. http://www.internet2.edu/sip.edu/ Both SPA-3000's and the routers that are connected to need to have a static WAN ip address OR so you can always know your ip address. Your routers should be configured to allow the SIP ports 5060 and 5061 and foeward these two on your SPA-3000's and should be "SIP friendly". Enable Make and receive calls without registration on both SPA-3000's and set the User ID and SIP Port on line 1 and PSTN line to unique numbers on both SPA-3000's. Enable Voip and PSTN gateways on both spa's, spa2 - Enable VoIP Caller Auth Method: PIN, Define the VoIP Caller 1 PIN: This PIN will be requested when you want to gain access to the pstn line. At this point from the spa1 you can dial #2 and the spa2 answer with some beeps (asking you for the PIN) then after you enter a correct pin, you will hear dial tone on the spa2. Adding a PIN would be a way to prevent incoming (or outgoing) calls without the user touch-toning some a secret key. If you wanted all calls transparently relayed to the other side, I'd think you wouldn't really want this. Also, from a security standpoint, unless the PIN is very long (say well over 6 digits), it isn't going to do much to stop a computerized attack. It just isn't going to take that long for a computer to try all 3, 4 or 5 number PINS. Trying all 6 number PINS probably only takes a week or two. To stop a determined attacker from making outgoing calls via your pbx you'll want to add md5/http-digest authentication to the internet side of the pbx-connected sipura. Just choose 32 randomly chosen letters and numbers for your md5 password. Load that password into both of your sipuras and you should be safe enough. One thing I forgot to mention is there are a number of forums where folks that like to hack their sipura's hang out. Most of them seem to have thought about this sort of stuff for much longer than I have. http://voxilla.com/PNphpBB2.html http://forum.sipbroker.com/search.php?searchid=4811 http://www.dslreports.com/forum/voip

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