To try to be a bit clearer: in receiving calls from the PSTN on (in this case) interface IF-FXO0 (which is bound to one fxo port) sometimes I get caller ID and sometimes I don't. When I don't something is creating a 'from' header of [email protected]:5060 . Some of these FXO's will go to doorphones and I'd like the callerID that shows up on the phones to be something like 'Lobby' or 'Dock' rather than 'anonymous'. I also have some POTS lines that do not receive caller ID and I'd like to identify which line if a call comes in on it ("AlarmCo" say).
I think I followed the outline of the setup here: http://sipx-wiki.calivia.com/index.php/Patton_4114_TESTED and if I understand correctly the called-e164 routing table is used to route calls from the SIP side toward the PSTN side (via the hunt group). There's no explicit table to route to the sip side, just the entry in the FXO interface 'route call dest-interface IF_SIPX'. Do I need to create a one entry routing table for pstn->sip calls just so I can add one of these maps to it? Or am I completely misunderstanding the purpose of the called-e164 table (does it apply in both PSTN->SIP and SIP->PSTN calls?). Thanks for your patience, -Eric Varsanyi On Jan 24, 2010, at 11:49 PM, Jim Canfield wrote: > I'm trying to understand what you are doing here. Sounds more like you > want to do address-translation on the from-header not create a mapping > table. The CID_MAP will do nothing until you use it somewhere. If > you want to use the CID_MAP just append it to the end of a routing > destination line. > > I've never tried to change URI's with mapping tables but I strip/add > digits all the time... > > Example: > > routing-table called-e164 TAB_IN > route default dest-service OUTBOUND > route *28.. dest-interface IF_PRI_2 REMOVE* > > mapping-table called-e164 to called-e164 REMOVE* > map *(.%)$ to \1 > > > > On Sun, Jan 24, 2010 at 9:21 PM, Eric Varsanyi <[email protected]> wrote: >> I have a Patton 4114 FXO box and its working well but I'd like to change the >> 'From' URI from 'anonymous' if no CID is received to something else (the >> depends on the FXO interface the call was received on). I tried creating a >> mapping table in the cs switch context: >> >> mapping-table calling-e164 to calling-uri CID_MAP >> map default to [email protected] >> >> However I still get From: [email protected] (where the nn's are the >> address of the patton). >> >> I suspect I'm just not getting how to attach this mapping table to the >> (implied?) routing table that handles called from the FXO to the SIP gateway >> in the call router. >> >> Any hints would be appreciated, I have 3 different classes of 'anonymous' >> I'd like to categorize based on which FXO port the call comes in on. >> >> Thanks, >> -Eric Varsanyi
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