FXS devices must call FXO. FXO devices must call FXS. If you have an FXS device and it is analog, you need a FXO gateway to receive it as FXO and convert it to sip, and vice versa.
Just in case it matters, they apartment building scenario is very doable, though I would not use sipx to backend that. We install door control systems, and "most' of our customers building anything today connect a telco (pots) line. These in turn call the tenants cell or home phone in the apartment. In instances where there is a dedicated intercom system or wiring for the door entry system, this can all be achived without a pbx. For simple door entry connecting to sip we would look at something like: http://www.cyberdata.net/products/voip/digitalanalog/intercom/index.html It's sip already, and poe. It connects as a user line to sipx and there the callerid is not an issue. It's for a single door. Any decent FXS gateway can be used (there are 24 port models out there) to convert an all analog system to call each other without a sipx server. No PBX would be needed, and can add a sipx server if/when needed, because it would simply be a gateway. I'm sure as the marketplace demands more sophisticated and sip ready door entry systems, you will find they will become available. I'm not sure the manufacturers are there yet. Valcom (www.valcom.com) makes a number of IP paging, paging servers, and other IP building communications products (clocks, door entry call buttons like the one above, speakers and horns). On Tue, Jan 26, 2010 at 10:00 AM, Eric Varsanyi <[email protected]> wrote: > The doorphone device provides talk battery and ring voltage, it IS an FXS > (it provides services to a station) and must be connected to an FXO port in > my infrastructure. If it was a little fancier it would provide callerid bell > style between the rings, but it isn't that fancy (it has no UI to enter a > callerID string for instance) hence the mapping in the Patton. > > Viking has FXO style doorphones too (that expect to hear a dialtone then > dial out), these would be more appropriate IMO in an apartment building or > rented suites scenario where you might want the person at the door to be > able to dial a specific apartment or house. > > Thanks again for the (weeks ago) tip on the Patton Smartnode line, they > really do rock. > > -Eric > > On Jan 26, 2010, at 6:42 AM, Tony Graziano wrote: > > > If they pretend to be a CO trunk, and simply dial a number, they want to > be > > connected to something that provides talk battery and dialtone. Just > pickup > > and dial, so they want an FXO port, not an FXS port. > > > > I don't see, if this worked on an FXS sipx would provide callerid for the > > user portion. > > > > What you are doing here is the correct method for an FXO port. The Patton > is > > the obvious choice to me, because it is highly configurable. > > ============================ > > Tony Graziano, Manager > > Telephone: 434.984.8430 > > Fax: 434.984.8431 > > > > Email: [email protected] > > > > LAN/Telephony/Security and Control Systems Helpdesk: > > Telephone: 434.984.8426 > > Fax: 434.984.8427 > > > > Helpdesk Contract Customers: > > http://www.myitdepartment.net/gethelp/ > > > > ----- Original Message ----- > > From: [email protected] > > <[email protected]> > > To: Eric Varsanyi <[email protected]> > > Cc: sipXecs users <[email protected]> > > Sent: Tue Jan 26 00:09:11 2010 > > Subject: Re: [sipx-users] Patton caller ID mapping > > > > OK, with the clue injection that I could use a routing table in interface > > FXO I came up with this (which works and transforms the no-callerid call > to > > a call from Lobby extension 270): > > > > interface fxo IF-FXO0 > > route call dest-table FXO0-TO-SIP > > disconnect-signal loop-break > > disconnect-signal busy-tone > > ring-number on-caller-id > > mute-dialing > > use profile tone-set US > > > > routing-table calling-e164 FXO0-TO-SIP > > route default dest-interface IF-SIP-PP FXO0-CID-FUNC > > > > complex-function FXO0-CID-FUNC > > execute 1 FXO0-CID-MAP-NAME > > execute 2 FXO0-CID-MAP-E164 > > > > mapping-table calling-name to calling-name FXO0-CID-MAP-NAME > > map ^$ to Lobby > > > > mapping-table calling-e164 to calling-e164 FXO0-CID-MAP-E164 > > map ^$ to 270 > > > > > > Instead of using a different routing table for each interface an > > optimization might be to have a single routing table which keys on called > > e164 (different doors ring different users in this case) and has a > mapping > > function for each one. This is the only way to do it if all doors ring to > > the same hunt group as far as I can tell. > > > > These Patton boxes are great, complicated and feature rich but logical > and > > easy to debug. > > > > -Eric Varsanyi > > > > On Jan 25, 2010, at 10:02 PM, Eric Varsanyi wrote: > > > >> These devices generate ring voltage and pretend to be an FXS. I thought > >> that sending ring voltage TO an FXS port would at best do nothing and at > >> worst blow it up (though unlikely unless both were glaring ring voltage > at > >> each other). > >> > >> These doorphones do not look for dialtone then dial out, they pretend to > >> be a CO trunk line so they look like an inbound call to your PBX (they > >> provide ring voltage and talk battery): > >> http://www.vikingelectronics.com/products/view_product.php?pid=268 > >> (W1000A). > >> > >> I'll try the (now obvious) method of just routing to a table instead of > an > >> interface then attaching the map to the table. Thanks! > >> > >> -Eric > >> > >> > >> On Jan 25, 2010, at 6:27 PM, Jim Canfield wrote: > >> > >>> On Mon, Jan 25, 2010 at 2:31 PM, Eric Varsanyi <[email protected]> > wrote: > >>>> Thank you very much for the response. > >>>> > >>>> I would think I could have a different route table for each IF_FXO > >>>> interface that uses a different mapping table, the devil (for me) is > >>>> that the example config doesn't have any route table at all for IF_FXO > >>>> calls, the FXO interface just points right at the SIP interface -- so > >>>> I'm not sure how to attach the route. > >>> > >>> Just point IF_FXO to the table. > >>> > >>> interface fxo IF_FXO0 > >>> > >>> route call dest-table YOUR_TABLE > >>> disconnect-signal loop-break > >>> disconnect-signal busy-tone > >>> ring-number on-caller-id > >>> dial-after timeout 2 > >>> mute-dialing > >>> use profile tone-set US > >>> > >>> ..there is no rule that says you have to direct to an interface. > >>> > >>>> > >>>> Now that I know I'm on the right track at least I'll study the docs > some > >>>> more (and probably ping Patton) and let you and the list know what I > >>>> come up with. > >>>> > >>> > >>> After thinking about this, it sounds like you have several in-house > >>> devices like 'Lobby' or 'Dock' It would be much cleaner if you used > >>> FXS ports and have them register as sip users rather than try and > >>> remap FXO ports. > >> > >> _______________________________________________ > >> sipx-users mailing list [email protected] > >> List Archive: http://list.sipfoundry.org/archive/sipx-users > >> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > >> sipXecs IP PBX -- http://www.sipfoundry.org/ > > > > _______________________________________________ > > sipx-users mailing list [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipx-users > > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > > sipXecs IP PBX -- http://www.sipfoundry.org/ > > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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