IP Phone Pickup PSTN Call

My * server has exceeded my expectations. I don’t use even 1/10th of what it could do so I have begun exploring more ways to work with it. But I’m not sure if this is do-able or not.

My setup has a single X101P card that takes calls from the PSTN line into my house and sends them into my dial plan. I a couple of IP phones and other analog phones around the house as well. When a call comes in * waits for 30 seconds before answering. I know that’s an eternity but it gives me a chance to answer the call on one of the analog phone first. If I don’t answer the caller gets a chance to ring one of the IP phones or just wait for voicemail.

My question is this. Is there a way to setup the dial plan so that a call could be answered by one of the IP phones as well. I know how to do this but that would mean having * answer the call right away which would mean that none of the anolog phones would get a chance to answer.

Here is a couple different ideas;

First solution: You could use ATAs with FXS ports and turn your analog phones into IP phones.

[ul]* All phones would be individually addressable for transfers, conference, ringing etc.

  • Depending on the ATA, you could still leave your regular line plugged into them to allow easier access to 911 etc. especially during a power outage.[/ul]
    [ul]* You’d need to purchase them and run Ehernet to all your phone locations. And don’t forget power needs as well.[/ul]

Second solution: Get FXS ports at your Asterisk box and plug all your phones into that, disconnecting them from direct POTS access. You may even get away with just one FXS port to plug them into (depends on many variables but I wouldn’t run too many phones on one port)

[ul]* No/little rewiring needed. Existing telephone line could be used. [/ul]
[ul]* Each phone wouldn’t be individually addressable unless they all have home runs and connect to individual FXS ports.

  • 911 maybe more difficult in a power outage if your asterisk box goes down. Check with your FXS/FXO card provider and see what happens in a power outage, and please make sure to TEST this as well!![/ul]

Third solution: (and you’re not going to like this one either), is to get a VOIP provider with DID service, get a new number (or port your existing if they can), and then have your callers call that number instead. Asterisk could then use the “followme” feature (that may not be the right term, can’t remember offhand) and ring your analog phones and ip phones at the same time. First one to pick up “wins” and gets the call. So;

DID VOIP call — Asterisk (ip phones ring) — VOIP Outbound call to old home number — analog phones ring.

Major cons with this are a lot more added expense with multiple VOIP provider calls costing you incoming and outgoing minutes, and the complexity/hassle of changing numbers etc. etc.

I use the “followme” feature in kind of a reverse scenario in that, my parents call my extension from their IP phone, it rings my IP phone at my house, and after 4 seconds of no answer it then starts ringing my cell phone too. Again, first device to pick it up wins. The second solution is probably the best one, but make sure to read and thoroughly understand the cons of this. 911 is very important feature that can easily be overlooked until it’s too late.

Good luck!


Deleting multiple posts.

What’s up with these forums not “submitting” posts correctly?! Drives me insane! Preview works fine, then the submit hangs forever. Try it again, hangs forever. Refresh window and looks like it didn’t post - then a few minutes later, multi-post hell shows up.


I think option #2 might be what I’m looking for. And if my (bong-soaked)memory serves me correct, someone once told me that I could plug my FXS port into one of the existing home jacks and all the other jacks in the house would have access to it. Did I rermember this right or do I need to take another toke?

That sounds workable, I just don’t know what would happen if you still had your incoming pots line hooked up at the same time. Never done it myself, so don’t know… probably not a good idea though, so I’d disconnect your incoming line from your analog phones and only have it feeding your FXO on Asterisk. Like this;

Incoming Pots ===> (FXO) Asterisk (FXS) <==== analog phones.


The forum posts everything correctly, its just that it is kind of slow when it is doing it and people click submit multple times thinking it will speed things up.
Maybe digium needs to look into it, or people need to just slow down