Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Am Samstag, den 15.12.2007, 16:55 -0800 schrieb Philip Prindeville: I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling number. This topic has been covered in length. In most cases it seems to end at the fact that providers correct caller-ids they get from the calling party: If you send any number which is assigned to the PRI (or SIP trunk), that is fine; if you send another number, it will be changed to the (first) number of the PRI/trunk. Few providers allow for foreign caller ids to be sent over their equipment - in some countries this is even illegal. For example, one of my providers (German) allows to set any CALLERID, but their documentation warns to not do stupid tricks, as calls can be tracked and using malicious information will be prosecuted. This feature is to be used only for sending _my_ cell phone number etc. BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Anselm Martin Hoffmeister wrote: In most cases it seems to end at the fact that providers correct caller-ids they get from the calling party: If you send any number which is assigned to the PRI (or SIP trunk), that is fine; if you send another number, it will be changed to the (first) number of the PRI/trunk. Few providers allow for foreign caller ids to be sent over their equipment - in some countries this is even illegal. For example, one of my providers (German) allows to set any CALLERID, but their documentation warns to not do stupid tricks, as calls can be tracked and using malicious information will be prosecuted. This feature is to be used only for sending _my_ cell phone number etc. Do you know of any GSM providers/contracts where faking for a valid reason is possible? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Philipp Kempgen wrote: Anselm Martin Hoffmeister wrote: In most cases it seems to end at the fact that providers correct caller-ids they get from the calling party: If you send any number which is assigned to the PRI (or SIP trunk), that is fine; if you send another number, it will be changed to the (first) number of the PRI/trunk. Few providers allow for foreign caller ids to be sent over their equipment - in some countries this is even illegal. For example, one of my providers (German) allows to set any CALLERID, but their documentation warns to not do stupid tricks, as calls can be tracked and using malicious information will be prosecuted. This feature is to be used only for sending _my_ cell phone number etc. Do you know of any GSM providers/contracts where faking for a valid reason is possible? Regards, Philipp Kempgen I can think of some... in rural Idaho, cell coverage is sparse. I might check my voice mail of my cell phone via a land line, and want to call back with a response originating from my cell number... Also the case if I have my cell set to forward-on-busy to my land line, or if I'm hosting an answering service, etc. -Philip ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Philip Prindeville wrote: Philipp Kempgen wrote: Do you know of any GSM providers/contracts where faking for a valid reason is possible? I can think of some... in rural Idaho, cell coverage is sparse. I might check my voice mail of my cell phone via a land line, and want to call back with a response originating from my cell number... Also the case if I have my cell set to forward-on-busy to my land line, or if I'm hosting an answering service, etc. What I'm looking for is this scenario: I call someone's cell phone number via my GSM gateway (to save money). But I'd like to set my landline number as the callerid (instead of one of the numbers of the GSM gateway or no callerid). Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling number. My dialplan looks like: [globals] ... TRUNK=SIP/sip_proxy-out CELL=${TRUNK}/208xxx PHILIP=SIP/bedroom_1SIP/office_2SIP/kitchen_1${CELL} [incoming] exten = s,1,Answer() ; sometimes signaling and media get out of sync on cell networks... exten = s,n,Wait(0.75) exten = s,n,Playback(main-menu) exten = s,n(exten),Background(vm-enter-number-to-call) exten = s,n,WaitExten(5) exten = s,n(goodbye),Playback(vm-goodbye) exten = s,n(end),Hangup ... exten = 111,1,Macro(stdexten,111,${PHILIP}) exten = 111,n,Goto(s,exten) exten = 112,1,Macro(stdexten,112,${REDFISH}) exten = 112,n,Goto(s,exten) ... exten = i,1,Playback(pbx-invalid) exten = i,n,Goto(s,exten) exten = t,1,Goto(s,goodbye) Ok, so far, so good. The problem is that when we hit Macro(stdexten,111,${PHILIP}) and it does the Dial(${PHILIP}) which includes the SIP/sip_proxy-out/208xxx, 208xxx rings with my PBX's extension. Oddly, the internal phones ring with outside caller's extension. [sip_proxy-out] type=peer fromuser=208nnn fromdomain=x.x.x.x host=y.y.y.y call-limit=5 nat=yes So I'm not setting the callerid on the peer by default. What am I missing? Do I need to modify the stdexten macro to dial with the 'o' option? Or can I set this explicitly with a 'Set' before calling the macro? Or do I need to be missing with the RDNIS? Oh, I'm running Asterisk 1.2.25... (yes, I'll upgrade when AstLinux upgrades). -Philip P.S. I tried adding |o to the end of the PHILIP variable, but this didn't seem to make a difference. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users