[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 ___ 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
Re: [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.
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Re: [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 ___ 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
Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
Andrew Kohlsmith wrote:
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.
Misunderstanding: I need to change the calleridnum because there is missing the
0 before the 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.
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.
That is set correctly and works between sip clients. it is only a problem when i
try to dial out over zap/g1.
-A.
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Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
Peter Svensson wrote: 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. Those options are only available in BRIStuff. ___ 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
Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
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${CALLERIDNUM}) or something?
That is set correctly and works between sip clients. it is only a problem
when i try to dial out over zap/g1.
Are you mangling the outoging caller ID in your Zap-terminating extension
contexts?
-A.
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Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
Peter Svensson wrote: On Mon, 8 Aug 2005, Kib Eki wrote: Hi, we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our old PBX. So now we could migrate to the * server. But, there are two things we can't live with: 1. A call from the outside to the old PBX is missing a leading 0 before the number. Ex: caller has number 0123456 - * routes to old pbx - old pbx sees 123456 as caller number. See internationalprefix, nationalprefix etc in the file zapata.conf. 2. A call made from a SIP client to the outside lacks the extension in the number: Ex: PSTN number is 6789-0. The extension 234 is not added to the PSTN number like 6789-234 when dialing out over the PSTN. Are you refering to the dialed number or the outgoing caller id (calling number)? yes i refering to the my outgoing number (caller id) Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
Andrew Kohlsmith wrote:
On Monday 08 August 2005 10:56, Kib Eki wrote:
Misunderstanding: I need to change the calleridnum because there is missing
the 0 before the number.
SetCIDNum(0${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.
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Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
On Mon, 8 Aug 2005, Eric Wieling aka ManxPower wrote: Peter Svensson wrote: See internationalprefix, nationalprefix etc in the file zapata.conf. Those options are only available in BRIStuff. They have been in HEAD for quite some time. The 1.0.x-releaes are note really usable in a lot of situations. Peter ___ 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
Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
On Mon, 8 Aug 2005, Kib Eki wrote: 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. That is set correctly and works between sip clients. it is only a problem when i try to dial out over zap/g1. Most likely you and your provider are not in agreement on how the calling party number should be encoded (number of digits and which Type Of Number / Numbering Plan). The TON/NPI is set with the prilocaldialplan option. Make sure you send the expected number of digits. You may have to do a SetCallerId() before the dial. Peter ___ 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
Re: [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${CALLERIDNUM}) or something?
Not to be nitpicky, but * will complain that SetCIDNum is deprecated.
SetVar(ZEROPREFIX=0)
Set(CALLERID(number)=${ZEROPREFIX}${CALLERIDNUM})
* will complain that SetVar is deprecated as well. But since I as yet
haven't found how to call the new SetVar, I just pretend I don't see
them. Anyone know?
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level,
I'm really quite busy.
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