Kib Eki
2005-Aug-08 01:03 UTC
[Asterisk-Users] Digium TE405P, caller id and migration to *
Hi, we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our old PBX. So now we could migrate to the * server. But, there are two things we can't live with: 1. A call from the outside to the old PBX is missing a leading 0 before the number. Ex: caller has number 0123456 -> * routes to old pbx -> old pbx sees 123456 as caller number. 2. A call made from a SIP client to the outside lacks the extension in the number: Ex: PSTN number is 6789-0. The extension 234 is not added to the PSTN number like 6789-234 when dialing out over the PSTN. Can anybody tell me how i must change the configuration? Do you need the zapata.conf? Thanks in advance and regards
Andrew Kohlsmith
2005-Aug-08 05:43 UTC
[Asterisk-Users] Digium TE405P, caller id and migration to *
On Monday 08 August 2005 04:03, Kib Eki wrote:> 1. A call from the outside to the old PBX is missing a leading 0 before the > number. Ex: caller has number 0123456 -> * routes to old pbx -> old pbx > sees 123456 as caller number.This is absolutely trivial to fix. Anyone who's been able to put * between a PRI and a PBX should be able to figure this out without asking the list. It's trivial dialplan stuff. exten => _X.,1,Dial(Zap/g2/0${EXTEN}) kind of trivial. You may have to debug a little to see where or why the 0's disappearing.> 2. A call made from a SIP client to the outside lacks the extension in the > number: Ex: PSTN number is 6789-0. The extension 234 is not added to the > PSTN number like 6789-234 when dialing out over the PSTN.Again, trivial dialplan stuff. Your sip.conf will have the callerid for each SIP client and you can append that information to the outgoing CID. -A.
Peter Svensson
2005-Aug-08 07:55 UTC
[Asterisk-Users] Digium TE405P, caller id and migration to *
On Mon, 8 Aug 2005, Kib Eki wrote:> Hi, > > we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our > old PBX. So now we could migrate to the * server. > > But, there are two things we can't live with: > > 1. A call from the outside to the old PBX is missing a leading 0 before the number. > Ex: caller has number 0123456 -> * routes to old pbx -> old pbx sees 123456 as > caller number.See internationalprefix, nationalprefix etc in the file zapata.conf.> 2. A call made from a SIP client to the outside lacks the extension in the number: > Ex: PSTN number is 6789-0. The extension 234 is not added to the PSTN number > like 6789-234 when dialing out over the PSTN.Are you refering to the dialed number or the outgoing caller id (calling number)? Peter
Kib Eki
2005-Aug-08 09:10 UTC
[Asterisk-Users] Digium TE405P, caller id and migration to *
Andrew Kohlsmith wrote:> On Monday 08 August 2005 10:56, Kib Eki wrote: > >>Misunderstanding: I need to change the calleridnum because there is missing >>the 0 before the number. > > > SetCIDNum(0${CALLERIDNUM}) or something?yes, but that does not work the zap channel connected the pbx. means i had no success with this> > >>That is set correctly and works between sip clients. it is only a problem >>when i try to dial out over zap/g1. > > > Are you mangling the outoging caller ID in your Zap-terminating extension > contexts?Yes.> > -A. > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >