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/ _______________________________________________ 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/
