IP Phone + Regular Phone - mixed environment

I’ll start by explaining what I plan to setup (just waiting on the hardware).
I’m waiting for my TDM400P w/ 1 FXO module and plan to connect my Cisco 7960 (SIP 7.4). I’ve only got 1 phone line in the house and was hoping that the IP phone could handle local calls like a regular phone would, but then route long distance calls out via SIPPhone or something.

My questions are:

  • What is Asterisk going to think of a mixed environment when there are 3 regular phones and an IP phone all sharing 1 line? I know people who have done this, but I really don’t know to what extent.

  • Based on the assumption that Asterisk won’t go nuts, if I put someone on hold with the 7960 and have hold music coming from Asterisk, will it hang the line up if a regular phone picks up elsewhere in the house?

  • What other weirdness should I expect in a mixed environment like this? I know I won’t be able to transfer calls between phones or anything like that…

Thanks!

[quote=“tflux”]- What is Asterisk going to think of a mixed environment when there are 3 regular phones and an IP phone all sharing 1 line? I know people who have done this, but I really don’t know to what extent.[/quote]it’s messy, but if you configure Asterisk to not answer it’s ok.

[quote=“tflux”]- Based on the assumption that Asterisk won’t go nuts, if I put someone on hold with the 7960 and have hold music coming from Asterisk, will it hang the line up if a regular phone picks up elsewhere in the house?[/quote] if someone else picks up while Asterisk is using the line it just becomes part of the audio IO … Asterisk has no clue what has happened.

[quote=“tflux”]- What other weirdness should I expect in a mixed environment like this? I know I won’t be able to transfer calls between phones or anything like that…[/quote] i would think the worst thing is the fact that Asterisk/Zaptel has no (or very little) concept of analogue line status … if you dial with the IP phone on the Zap channel, Asterisk will report it as answered, even if you end up joining a call that someone is making using another phone.

in your position i would probably get a LinkSys PAP2 and a FXS module for the card … then connect all the additional phones in as Asterisk extensions.

Well, it sounds like the zaptel drivers don’t keep an eye on voltage on the FXO modules, which is really the only way to detect additional phones being part of a call. I suppose with what Asterisk is to be used for, these mixed situations are not all that common :smile:

quite. except that lots of people dabble with Asterisk and use this kind of setup before they have the confidence/hardware to route everything through Asterisk.

I fit the description that baconbuttie put on, dabbling with * but not confident in it. As well, I agree with his line saying: wife => Background(“i just want a phone !”)

What I’ve found using a mixed environment is that it seems to work OK. * is unaware that a regular phone has picked up the call and joins it into the audio stream without a hiccup. Same applies if I jump in with a SIP or IAX soft phone.

The problem I’m experiencing is that * can’t tell the difference between in-use voltage and a ring event. So, when my tivo dials out at 2 am * decides to ring my ring-group (including my FXO phones) like there’s an incomming call.

The first (and last time) that happened, I moved my “hard” phones back to the hardline.

I have a setup at home that is somewhat similar to the one you are thinking of, that I have been using for the past 4 months…
I have * running on a Soekris net4801 single board computer. Of course, the computer case is too small (it’s slightly larger than a dual CD case) to take a an FXO card. So, I used a Linksys SPA-3102 box. Here is how my hardware is set up…

    • connected to 802.11g main home router, behind a firewall.
  1. One SIP hardphone (GXP-2000) connected to a 2nd wireless router that is networked with the main router.
  2. A Linksys (formerly Sipura) SPA-3102 VOIP-PSTN gateway connected to the 2nd router. It has 1 FXO and 1 FXS port. [cost:$90]
    The FXO port side goes into my telephone wall socket. The FXS is connected to an analog cordless phone.
  3. All other telephone wall sockets have Plain Old Telephones [POTs] (corded and cordless) plugged in.
  4. All incoming PSTN calls ring the POTs. Simultaneously, the SPA-3102 routes SIP invites to my SIP phones through * WITHOUT taking the line off the hook. Thus, I can answer using either a SIP phone or a POT, transparently.

My observations:

  1. The SPA-3102 gateway does not seem to care if a POT is taken off-hook while a SIP-PSTN call is in progress (through itself). That’s good because this is “expected” behavior on a traditional, completely analog setup.
  2. Should a 2nd call come in when the first is in progress between SIP-PSTN through the SPA, then call waiting and call-waiting id are passed on to the SIP phone and the FXS line just fine. So that’s great! But as expected, the POT instruments don’t ring. No problem there either …expected behavior.
  3. But what I love most about the setup is (a) the ability to zap telemarketers selectively… I have an elaborate dialplan to get around their various CID tricks, while passing through legit numbers, and (b) I can use the * database to store and look up cid names, in case the cid name doesn’t come through, or if the cid name comes up as a “wireless caller”, or some such.

I only reboot when I make significant updates to the * setup, and I haven’t really had to reboot my * box, or the SPA or the routers for any other reason. I have noticed I have gone as long as 2 months between reboots without any problems.

I suppose, ymmv depending on the FXO card/box that you use. My experience with the SPA-3102 has been overwhelmingly positive. The only downside is that Linksys/Sipura have very little documentation for it. I had a devil of a time getting it to work correctly at first, but after the initial learning curve, it’s been largely trouble-free and easy to administer.

My setup includes two analog phones connected to the same PTSN line as the FXO module on the digium card in my server. I have no problems with this setup. As noted earlier, the key in this type of setup is not have asterisk “answer” the PSTN call in the dialplan. When asterisk detects a PSTN call on the ZAP channel, I dial the phone’s connected to the server I want to ring for 30 seconds. This gives me time to answer it with one of the two analog phones connected on the line or one of the asterisk phones I am dialing. I can answer on any of them without any problems, other than loss of cool asterisk features if I answer with one of the analog phones. If the call is not answered within the 30 seconds, then the dialplan sends the call to voicemail. The zap channel is answered and the caller hears my unavailable voice mail greeting.

I do not have a problem with asterisk detecting ring, when I grab one of the analog lines to make a call. There is a message about the zap channel on the console, but it is not causing any dialplan contexts to be executed.

It sounds like I have a similiar setup to SuperB. My * box is using 1 wildcard connected to the house PSTN, with a couple softphones spread across my network. My PSTN also has distinctive ring; 1 number is the house line the other number is fax/my test line.

I made all the settings to distinquish the 2 ringtones and it works great. However, lately we have been having an issue. When a call is placed from one of the regular house phones * sees that as an incoming call on the 2nd ringtone!

Any thoughts on how to stop/get around this?

I am hoping to do the exact same thing with my setup. I’ll have a few IP phones running through Asterisk, but the rest of the house phones and analog answering machine will be completely separate.

Can anyone point me in the right direction to configure asterisk to not “answer” calls but still ring my SIP phone?

Thanks for the help – it’s much appreciated.

I’ll add my 2 cents. I’ve had a mixed environment setup for a year or so. When a PSTN call comes in, as others have mentioned, it rings the POTS phones and * rings my voip phones. (I’m using a TDM card) My observations:

  1. If I answer from a voip phone, I then yell to the rest of the house that the call is for them, but make sure I hang up the line when the wife picks up a POTS phone. I can even put the call on hold, which plays music, but make sure I hang up * or the caller hears music and my wife screaming about the stupid phone system! Ha!

  2. If my wife answers a POTS phone and puts the call on hold, I have to run to another POTS phone to pickup. I have not found a way to pick up the line from within asterisk. Anybody else figure that out?

  3. I have played with the time that * rings the voip phones. Problem is * keeps ringing the phones even after a POTS phone has picked up. I have * set to only ring for 10 seconds or so. Any longer than that and it is really annoying to have the voip phones ringing 15 seconds after someone has answered from a POTS phone.