Re: [asterisk-users] Caller ID (name) - where does it come from?
For the CNAM vendors who pride themselves on completeness/coverage, don't you think that they have some interest in getting data from the likes of Teliax? Maybe they wouldn't pay for it, but ITSPs have to realize that to retain certain customers that they have to their customers numbers disseminated. But I guess if they can charge extra for what used to be table stakes, so be it. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Tuesday, July 07, 2009 8:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID (name) - where does it come from? My primary issue is for calls that are placed FROM my client's PBX, via VOIP provider (Teliax). The recipients of those calls are the ones that are not getting the proper CNAM information as the call comes in. I neglected to go into detail on this point at the end of my last post because I thought it was out of scope. But now that you ask... While I am not an expert in the specific architecture of CNAM database, I do know that (to Frank's point) it is not at all a database in the 'MySQL' or 'Oracle' sense of the word. It's a database more analagous to the DNS where data can be located in many places (cached) but there is a single source considered authoritative that ultimately propogates out to cache. This authoritative source is the Telco that provides your DID number--after all, they the only ones with a billing relationship to validate the name information. So historically, *normally* your Telco is the authoritative source of the CNAM data that populates the 'screens' of the people you call, and *normally* the Telco of the calling party is ultimately compensated by the Telco of the called party for providing the CNAM data, but this model has broken down in the world if IP telephohy. Your ITSP (Teliax) is one of them-thar new-fangled ITSPs and the big boys have exactly ZERO interest in compensating them for CNAM dips. Meanwhile they are excluded from the holy brotherhood of 'real' CNAM. This is why your name is not populated in the CNAM database. Teliax is not one of the CNAM insiders who exchange name data and compensate each other for said data. That's also why it would never make sense to ask your CNAM lookup serive provider to make corrections to errant CNAM data. It just doesn't work that way. It used to be that you could work around this problem by using LNP to port your number temporarily to an ILEC . Your TN would get a CNAM record which would persist as an orphan for years. Recently this has changed, and NOW when you port your TN away from the losing LEC, they purge your CNAM record. :-( Recently there are some good solutions to this problem. One is to ask your ITSP if they can put your number in the LIDB for a fee or alternatively you can just buy a white pages entry (also from your ITSP) which accomplishes the same thing. I've seen this for $5 per month, and the BONUS you get a white pages entry (which you may or may not want). I hope this helps. -Karl http://www.hcst.net/ 937-427-9000 ___ -- 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 ___ -- 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 ___ -- 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] Caller ID (name) - where does it come from?
How does that work? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Tuesday, July 07, 2009 8:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID (name) - where does it come from? snip I get paid every time I call someone that subscribes to caller ID. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 ___ -- 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] Caller ID (name) - where does it come from?
So how does Teliax (for instance) go about getting their client's information into these directories? Do they establish a relationship with someone like TargusInfo (described above)? How do other ITSP's provide this service, or do they ignore it as well? On Tue, Jul 7, 2009 at 9:49 PM, Frank Bulk frnk...@iname.com wrote: How does that work? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Tuesday, July 07, 2009 8:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID (name) - where does it come from? snip I get paid every time I call someone that subscribes to caller ID. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 ___ -- 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 -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- 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] Caller ID (name) - where does it come from?
Barry D. Hassler wrote: So how does Teliax (for instance) go about getting their client's information into these directories? Do they establish a relationship with someone like TargusInfo (described above)? How do other ITSP's provide this service, or do they ignore it as well? Yes, either they will work with Targus, Versign, etc who do commercial CNAM hosting, or they work with their CLEC partner who would presumably already have such an agreement in place. It's my understanding that with an LOA from your CLEC or ILEC (kind of the opposite of the LOA you need to port a number), you can have your own CNAM records hosting with one of the companies listed above and make money. We're talking a fraction of a cent per call so unless you have many DIDs and many more calls, it's not usually worthwhile. Most smaller ITSPs either don't know how this or don't have the volume to make it feasible. In Canada, we just include the name in the SS7 signaling on a per-call basis and bypass this whole mess :) Best regards, -- Trevor Peirce Digital Conceptions Canada http://www.digitalcon.ca 1-888-606-3030 / 250 483-0386 ___ -- 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] Caller ID replacement
The hotdesking section of the asterisk book may also be of interest... d 2009/2/13 David Ruggles da...@safedatausa.com Some googling lead me to this: http://hans.fugal.net/blog/tag/astdb Which looks like it has an answer. Thanks all! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Thursday, February 12, 2009 12:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Caller ID replacement Could you give me an example of how this would look in the dialplan? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Nicholson Sent: Thursday, February 12, 2009 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID replacement On Thu, 2009-02-12 at 11:27 -0500, David Ruggles wrote: I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. I would implement this using the a database (astdb or odbc) containing the mapping from cell number to extension. Then for each call that may need callerid modification, you can check the database for the proper mapping. With this method it is also easy to add new mappings. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1936 - Release Date: 02/05/09 11:34:00 ___ -- 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1936 - Release Date: 02/05/09 11:34:00 ___ -- 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 ___ -- 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] Caller ID replacement
I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. Any suggestions? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- 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] Caller ID replacement
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 David Ruggles wrote: global variables that link the cell phone #'s and extensions and have this done somewhat automagically. Load your cross-reference in AstDB and do the lookup that way. If the cell number exists in the database, replace the callerID with the extension number. If it doesn't exist then it must be from someone else so don't change the callerId. BK -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJlFGwCFu3bIiwtTARAlRSAJ48FS53xS4u0eIeJ63VrZulPZxMMQCffFHw 7riqdRkR6vq5tGT9Z78FpiQ= =SuKH -END PGP SIGNATURE- ___ -- 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] Caller ID replacement
On Thu, 2009-02-12 at 11:27 -0500, David Ruggles wrote: I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. I would implement this using the a database (astdb or odbc) containing the mapping from cell number to extension. Then for each call that may need callerid modification, you can check the database for the proper mapping. With this method it is also easy to add new mappings. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- 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] Caller ID replacement
Could you give me an example of how this would look in the dialplan? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Nicholson Sent: Thursday, February 12, 2009 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID replacement On Thu, 2009-02-12 at 11:27 -0500, David Ruggles wrote: I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. I would implement this using the a database (astdb or odbc) containing the mapping from cell number to extension. Then for each call that may need callerid modification, you can check the database for the proper mapping. With this method it is also easy to add new mappings. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1936 - Release Date: 02/05/09 11:34:00 ___ -- 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] Caller ID replacement
Some googling lead me to this: http://hans.fugal.net/blog/tag/astdb Which looks like it has an answer. Thanks all! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Thursday, February 12, 2009 12:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Caller ID replacement Could you give me an example of how this would look in the dialplan? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Nicholson Sent: Thursday, February 12, 2009 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID replacement On Thu, 2009-02-12 at 11:27 -0500, David Ruggles wrote: I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. I would implement this using the a database (astdb or odbc) containing the mapping from cell number to extension. Then for each call that may need callerid modification, you can check the database for the proper mapping. With this method it is also easy to add new mappings. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1936 - Release Date: 02/05/09 11:34:00 ___ -- 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1936 - Release Date: 02/05/09 11:34:00 ___ -- 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] caller ID - handle_request_invite: Failed to authenticate user
Joseph wrote: We have a caller ID from our phone provider Shaw Cable (digital phone) and it was working OK until recently. I get an error: WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have 4, digest has pstn- NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user THELMA sip:7804789...@10.10.0.103;tag=50e17675d59121c4o1 at this point call fails, it is not being passed through to asterisk. I'm using Linksys 3102, PSTN answer delay is set to 3sec. to allow for caller ID to pass through. When I decrease timing to 1sec. or eliminate it 0sec the call goes through but there is no caller ID being forwarded. It was working OK for a while. So I'm not sure if Shaw Cable have upgraded something on their digital phone or there is a problem with asterisk/ 4 is a Line1 pstn- is PSTN Line Have you tried to extend that delay to 5 or 6 seconds? It's possible that caller id is being sent a second or two later/longer, but your 3 seconds is now cutting off a portion of the caller id data. Lyle ___ -- 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] caller ID - handle_request_invite: Failed to authenticate user
We have a caller ID from our phone provider Shaw Cable (digital phone) and it was working OK until recently. I get an error: WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have 4, digest has pstn- NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user THELMA sip:7804789...@10.10.0.103;tag=50e17675d59121c4o1 at this point call fails, it is not being passed through to asterisk. I'm using Linksys 3102, PSTN answer delay is set to 3sec. to allow for caller ID to pass through. When I decrease timing to 1sec. or eliminate it 0sec the call goes through but there is no caller ID being forwarded. It was working OK for a while. So I'm not sure if Shaw Cable have upgraded something on their digital phone or there is a problem with asterisk/ 4 is a Line1 pstn- is PSTN Line -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] Caller ID sip trunk
Hi
I'm using the latest 1.4 asterisk, when I get an incoming call from
sipgate ( my only sip trunk) the variable Noop(${CALLERID(num)})
is populated with the ower channel ID not the callerid is this correct?
the correct callerid show on the internal phones though!!
if so how do I get the callerid from an incoming sip trunk?
Thanks for your help
Robb
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[asterisk-users] Caller ID service and the ethernet stucking
Hi All; We added the callerid service on our telephone line, once that done, now when we call to the Asterisk PBX or we need to place outside call via the digium (zaptel channel), the PBX got a problem in the network, and we become not able to reach it, this stay for a while of time (about 5 min) and then it come back reachable. I did not do any thing when the callerid service added by the telecom service provider, and I am surprised why this callerid service effect on the ethernet port? Did any one face this problem? My asterisk version: 1.4.19.2 My zaptel version: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10.1 Zaptel Echo Canceller: MG2 INFO-xpp: FEATURE: with sync_tick() from ZAPTEL In the /var/log/asterisk/messages, I did not find any message that help (warning or error). Any advise? Did any one face such problem? The PBX located in KSA. Regards Bilal ___ -- 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] Caller ID in IAX trunk, SIP trunk, between extensions and from FXO
Hi All; If I need to see on my Polycom LCD the caller id of the other caller extension (for example, if 801 called the polycom of 802 then how can I let the LCD of polycom of the extension 802 to display the 801 as caller)? My polycom model is 330. Also, I have IAX trunk between two Asterisk boxes, in the beginning I was able to see the telephone number on the Polycom 330 LCD of the caller number from other Asterisk, (I was able to see the mobile number that was calling to Asterisk Box A and enter the extension of the Polycom that is registered with Asterisk Box B), but now, I am not able. So what the missing configuration that might cause this problem? From the other side, I have also an SIP trunk between my Asterisk Box and a SIP softswitch, how can I set the caller id of my Asterisk to see it on the SIP softswitch (for example, to be the same as the extension that placed the call, or even to be a fixed number). Any advise? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller-id on X100P fails frequently
On Thu, May 15, 2008 at 5:07 PM, Daniel Lynes [EMAIL PROTECTED] wrote: Brian J. Murrell wrote: I have a Wildcard FXO: Wildcard X100P (clone) in my Asterisk (1.4.17) machine and as of late, Caller-ID on it seems to be failing more frequently than not. Sometimes I get callerid.c:613 callerid_feed: Caller*ID failed checksum sometimes it fails without even that. Also, try putting a one second wait() at the first priority in the extension. I have had two X100Ps running in the same box for years and they never miss CID when it's available. I think there were issues without using wait() though. ___ -- 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] caller-id on X100P fails frequently
I have a Wildcard FXO: Wildcard X100P (clone) in my Asterisk (1.4.17) machine and as of late, Caller-ID on it seems to be failing more frequently than not. Sometimes I get callerid.c:613 callerid_feed: Caller*ID failed checksum sometimes it fails without even that. In Zapata.conf I have: usecallerid=yes cidsignalling=bell cidstart=ring I'm in Bell Canada land if that makes any difference. Any ideas on how to make it more reliable? b. signature.asc Description: This is a digitally signed message part ___ -- 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] caller-id on X100P fails frequently
Brian J. Murrell wrote: I have a Wildcard FXO: Wildcard X100P (clone) in my Asterisk (1.4.17) machine and as of late, Caller-ID on it seems to be failing more frequently than not. Sometimes I get callerid.c:613 callerid_feed: Caller*ID failed checksum sometimes it fails without even that. It can fail for one of three reasons on an X100P card: 1. The caller blocked their caller ID 2. Your gains are not set correctly in zapata.conf (I believe the one for caller id is rxgain). 3. You're using an X100P card...the caller ID hardware on it has always had problems, and some X100P/X101P cards are worse than others. Any ideas on how to make it more reliable? Try adjusting your gains, or replacing the card. ___ -- 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] Caller id issue and Dial tone for sip phone on zero dialing
Hi all, I am not getting the dial tone when i dial the zero digit. And i am using analog card,for my operator phone caller id is not displaying on the phone.I am in india. In india is it possible to get the caller id for analog cards. Can any body help me. Please reply. ThanksRegards, sandeep.s___ -- 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] caller id issue for INDIA
hi all, how to set the caller id facility for the TDM400p card in INDIA. thanks sandeep.s ___ -- 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] caller id issue for INDIA
Hi, For the caller id there is a patch available for digium cards. you can patch that file. I am not aware about those files. so please refer some googleing. On Jan 18, 2008 2:57 PM, sandeep [EMAIL PROTECTED] wrote: hi all, how to set the caller id facility for the TDM400p card in INDIA. thanks sandeep.s ___ -- 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 -- Thank you with regards, Gopal, ___ -- 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] caller id issue for INDIA
On Fri, Jan 18, 2008 at 02:57:23PM +0530, sandeep wrote: hi all, how to set the caller id facility for the TDM400p card in INDIA. http://bugs.digium.com/6683 Hmmm looks like it needs some love and care. I wasn't following it carefully. Can anybody update me on it? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Caller ID Issue
I have a strange issue with CLID that I would appreciate if someone could point me in the right direction. When a call comes in (either from another SIP user on the same Asterisk box or from the ISDN PRI) the Caller ID Name is displayed correctly, but the Caller ID Number seems to be empty. My Grandstream phone is setting the Caller ID number to the registered account name while SJ Phone soft client shows the Caller ID number as empty. Any suggestions would be greatly appreciated. ___ --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] Caller ID Issue
Am Mittwoch, den 12.12.2007, 09:14 -0500 schrieb Lutgring, Sam:
I have a strange issue with CLID that I would appreciate if someone
could point me in the right direction. When a call comes in (either
from another SIP user on the same Asterisk box or from the ISDN PRI)
the Caller ID Name is displayed correctly, but the Caller ID Number
seems to be empty. My Grandstream phone is setting the Caller ID
number to the registered account name while SJ Phone soft client shows
the Caller ID number as empty.
Any suggestions would be greatly appreciated.
Hi Sam,
some phones seem to hate phone numbers with strange characters in them;
those might be spaces, + signs, - dashes etc. and refuse to display
anything at all. Perhaps the information is there, but it is in some way
or another taken as invalid.
You could see what Asterisk thinks those variables are. A
NOOP(CALLER-ID-Info: ${CALLERID(num)} / ${CALLERID(name)})
in the dialplan, together with CLI and set verbose 10 should show you
lots of information.
BR
Anselm
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Re: [asterisk-users] Caller ID Question
That definitely makes sense, but how is it done on the phone level (polycom 501 and 601s)? I looked through all the configs and can't seem to find it in there, which is why I thought it might be an asterisk thing. Rob CunningPike wrote: Disable URI dialing on your phones. CP Rob Schall wrote: I have an asterisk 1.4 setup with a PRI installed and working. We are using a Polycom 501 to test the setup.. Inbound calls work great as do phone to phone calls. However in all cases, the caller id is a bit odd. It shows: 99 sip:[EMAIL PROTECTED] what cause's this? How do I get just 99? Thanks ___ --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 ___ --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 ___ --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] Caller ID Question
Rob Search your sip.cfg in your polycom directory for feature.9.name=url-dialing feature.9.enabled=1 Set it to enabled=0 I have this same issue with my 550's and 650's Turning off URI dialing in the Polycoms only fixed the inbound calls all internal cals still show [EMAIL PROTECTED] I am learning to ignore it Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Monday, November 26, 2007 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID Question That definitely makes sense, but how is it done on the phone level (polycom 501 and 601s)? I looked through all the configs and can't seem to find it in there, which is why I thought it might be an asterisk thing. Rob CunningPike wrote: Disable URI dialing on your phones. CP Rob Schall wrote: I have an asterisk 1.4 setup with a PRI installed and working. We are using a Polycom 501 to test the setup.. Inbound calls work great as do phone to phone calls. However in all cases, the caller id is a bit odd. It shows: 99 sip:[EMAIL PROTECTED] what cause's this? How do I get just 99? Thanks ___ --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 ___ --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 ___ --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] Caller ID Question
I am having this problem as well. Disabling URL-dialing in sip.cfg has worked in previous polycom firmware versions. However on the latest firmware 2.2.0 disabling url-dialing does not work.. Has anyone else hove it working in ver 2.2.0? Thanks Doug Gillespie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Bell Sent: Monday, November 26, 2007 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID Question Rob Search your sip.cfg in your polycom directory for feature.9.name=url-dialing feature.9.enabled=1 Set it to enabled=0 I have this same issue with my 550's and 650's Turning off URI dialing in the Polycoms only fixed the inbound calls all internal cals still show [EMAIL PROTECTED] I am learning to ignore it Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Monday, November 26, 2007 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID Question That definitely makes sense, but how is it done on the phone level (polycom 501 and 601s)? I looked through all the configs and can't seem to find it in there, which is why I thought it might be an asterisk thing. Rob CunningPike wrote: Disable URI dialing on your phones. CP Rob Schall wrote: I have an asterisk 1.4 setup with a PRI installed and working. We are using a Polycom 501 to test the setup.. Inbound calls work great as do phone to phone calls. However in all cases, the caller id is a bit odd. It shows: 99 sip:[EMAIL PROTECTED] what cause's this? How do I get just 99? Thanks ___ --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 ___ --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 ___ --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 ___ --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] Caller ID Question
Yep, I disabled it in my sip.cfg and everything worked like a charm. :) Rob asterisk wrote: I am having this problem as well. Disabling URL-dialing in sip.cfg has worked in previous polycom firmware versions. However on the latest firmware 2.2.0 disabling url-dialing does not work.. Has anyone else hove it working in ver 2.2.0? Thanks Doug Gillespie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Bell Sent: Monday, November 26, 2007 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID Question Rob Search your sip.cfg in your polycom directory for feature.9.name=url-dialing feature.9.enabled=1 Set it to enabled=0 I have this same issue with my 550's and 650's Turning off URI dialing in the Polycoms only fixed the inbound calls all internal cals still show [EMAIL PROTECTED] I am learning to ignore it Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Monday, November 26, 2007 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID Question That definitely makes sense, but how is it done on the phone level (polycom 501 and 601s)? I looked through all the configs and can't seem to find it in there, which is why I thought it might be an asterisk thing. Rob CunningPike wrote: Disable URI dialing on your phones. CP Rob Schall wrote: I have an asterisk 1.4 setup with a PRI installed and working. We are using a Polycom 501 to test the setup.. Inbound calls work great as do phone to phone calls. However in all cases, the caller id is a bit odd. It shows: 99 sip:[EMAIL PROTECTED] what cause's this? How do I get just 99? Thanks ___ --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 ___ --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 ___ --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 ___ --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 ___ --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] Caller ID Question
We just installed an Asterisk 1.4 system and have a Polycom 501 phone we are using to test it. We have a PRI installed as well and it works well. The problem When a call is incoming, the caller id says: 99 sip:[EMAIL PROTECTED] how do you get it to just say 99 and remove all of the rest? Thanks ___ --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] Caller ID Question
I have an asterisk 1.4 setup with a PRI installed and working. We are using a Polycom 501 to test the setup.. Inbound calls work great as do phone to phone calls. However in all cases, the caller id is a bit odd. It shows: 99 sip:[EMAIL PROTECTED] what cause's this? How do I get just 99? Thanks ___ --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] Caller ID Question
On Wed, 21 Nov 2007 12:54:22 -0600, Rob Schall [EMAIL PROTECTED]
wrote:
what cause's this? How do I get just 99?
Maybe there's a better way, ie. making the ISDN card or Polycom unit
handle the presentation, but you could have Asterisk rewrite the CID
name/number on the fly.
${CALLERID(num)})
${CALLERID(name)})
${DB(cidname/${CALLERIDNUM})})
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Re: [asterisk-users] Caller ID Question
I have the same issue and I cant fix it :(
On Nov 21, 2007 9:56 PM, Vincent [EMAIL PROTECTED] wrote:
On Wed, 21 Nov 2007 12:54:22 -0600, Rob Schall [EMAIL PROTECTED]
wrote:
what cause's this? How do I get just 99?
Maybe there's a better way, ie. making the ISDN card or Polycom unit
handle the presentation, but you could have Asterisk rewrite the CID
name/number on the fly.
${CALLERID(num)})
${CALLERID(name)})
${DB(cidname/${CALLERIDNUM})})
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Re: [asterisk-users] Caller ID Question
Are you calling the other phones by URL or through asterisk? if your phone is registered to asterisk, and you ask to dial a number, it will connect through asterisk to another registered phone. If you ask to dial a url from the polycoms, i.e. sip:[EMAIL PROTECTED], then it will connect directly to the other SIP UA, skipping asterisk entirely. This is typically when you see the URL on the screen of the receiving phone. Am I clear? Sorry if I'm not. Mojo Rob Schall wrote: I have an asterisk 1.4 setup with a PRI installed and working. We are using a Polycom 501 to test the setup.. Inbound calls work great as do phone to phone calls. However in all cases, the caller id is a bit odd. It shows: 99 sip:[EMAIL PROTECTED] what cause's this? How do I get just 99? Thanks ___ --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 ___ --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] Caller ID Question
Disable URI dialing on your phones. CP Rob Schall wrote: I have an asterisk 1.4 setup with a PRI installed and working. We are using a Polycom 501 to test the setup.. Inbound calls work great as do phone to phone calls. However in all cases, the caller id is a bit odd. It shows: 99 sip:[EMAIL PROTECTED] what cause's this? How do I get just 99? Thanks ___ --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 ___ --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] Caller ID on Channelized T1 (EM Wink)
Hi, Normally my T1 implementations are PRI. However, I do have a customer who uses channelized T1 (24 channels). I have setup a 'test' environment, and have two T1 channels back-to-back in my [*] box. Both are setup with signalling = em_w. Calls DO go back forth, but I can not see the callerID being passed. Any ideas? WW -- Willy Wouters, PhD Asterisk Telephony Web Applications MAGU ENTERPRISES Tel: 713-474-1534 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Caller ID on Channelized T1 (EM Wink)
I've not seen an EM/Wink that supported Caller*ID. You can fake it by sending something like *CALLERID*DID and then on the far end break that out and set the callerid and goto the DID. Willy Wouters wrote: Hi, Normally my T1 implementations are PRI. However, I do have a customer who uses channelized T1 (24 channels). I have setup a 'test' environment, and have two T1 channels back-to-back in my [*] box. Both are setup with signalling = em_w. Calls DO go back forth, but I can not see the callerID being passed. Any ideas? WW ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] caller ID strangeness
when executing a NOOP(caller id ${CALLERIDNUM}) in the dialplan
I am getting odd caller id results from a SIP connection. The SIP
Connection is to
a nortel cs 1000.
*4145664222;phonecontext=+1
notice the extra stuff after the number
I am using asterisk 1.2.17
Is there a caller ID issue?
Jerry
*
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Re: [asterisk-users] caller ID strangeness
--- Jerry Geis [EMAIL PROTECTED] wrote:
when executing a NOOP(caller id ${CALLERIDNUM})
I am using asterisk 1.2.17
I use CALLERID(num) or CALLERID(all) in 1.2+.
I don't know if that can help.
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Re: [asterisk-users] Caller ID Spoofing to be banned in the USA
You are right but my concerns is the ITSP's may stop allowing it because they don't want to get in to trouble. They may request a list of all the DID's that I have and limit me setting my CID to the list that I gave them. - Original Message - From: Andrew Joakimsen To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, July 03, 2007 6:20 AM Subject: Re: [asterisk-users] Caller ID Spoofing to be banned in the USA The Proposed bill S704 reads It shall be unlawful for any person within the United States, in connection with any telecommunications service or IP-enabled voice service, to cause any caller identification service to transmit misleading or inaccurate caller identification information, Please tell me how you can construe making a call with the the CID of a number in your control to be Misleading or inaccurate On 7/2/07, Dovid B [EMAIL PROTECTED] wrote: Anyone know if this is only to bother some one ? I have a client that has a consulting business. The clients call in and his asterisk server call's his cell when he is out of the office. It passes along the CID. I hope the laws don't screw this up for those that change CID on every call for legitimate reasons. - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, June 28, 2007 5:43 PM Subject: [asterisk-users] Caller ID Spoofing to be banned in the USA Anyone running caller id spoofing applications in the USA running asterisk? Then it's time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-to-be-outlawed.html Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -- ___ --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 ___ --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 -- ___ --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___ --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] Caller ID Spoofing to be banned in the USA
Dovid B wrote: You are right but my concerns is the ITSP's may stop allowing it because they don't want to get in to trouble. They may request a list of all the DID's that I have and limit me setting my CID to the list that I gave them. I doubt this will ever be an issue. The telco companies and certainly not one for major changes, nor are they likely to enforce any laws and put any time into something they aren't responsible for. My understand of this new law is the punish those users who abuse the ability to set the CID. If the government was really bored and wanted to waste a lot of time and money in court, I guess they could try to take ATT or Broadwing to court, but my guess is they wouldn't get far. It isn't their fault that a customer of theirs broke the law, and they haven't (and probably won't) be required to keep people from breaking the law. Just because you make and sell guns doesn't mean you can go after them for using your guns and bullets to kill people. Ya know? Long story short, unless the government has nothing else to go after, I'd think you'll be perfectly fine. Like others have said on this subject... Unless you do it to fool somebody and it has a mean spirit to it, you should be in the clear. Masking a caller-id to be a cell phone of an employee, etc, should be fine, as long as every one knows what is going on. Rob ___ --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] Caller ID Spoofing to be banned in the USA
On Jul 2, 2007, at 11:56 PM, Ron Stephan wrote: Please tell me how you can construe making a call with the the CID of a number in your control to be Misleading or inaccurate Sure - it goes like this - The less scrupulous among us might use a spoofed cid to get people to do something they normally wouldn't. Imagine a spoofed CID of your corporate headquarters and somebody calling your employees saying they were HR and needed to confirm SSN numbers... I will let you fill out the rest of the disaster. Trouble is you don't think as evil as some people do. It annoys the hell out of me too - I would love to spoof my cell CID. I would love to have three or four cells with the same CID (all pointing back to my astericks box). It seems damn near impossible hear in Kalifornia. Ron Elvis Stephan Not to not pick, but I think you went beyond what Andrew was saying... Misleading or inaccurate, I would read this to imply that I'm not miss leading you, I'm not providing inaccurate information I am providing you with a means to contact me back. I'm opening stating who I am and where you can reach me with no malitious attempt. The Misleading or inaccurate part would encompass the scenario you describe above. Much the same the intention isn't to target corporate offices where employee's have DID's but their caller ID shows up as the trunk line which feeds to the building / company operator. Now, if I goto a provider and tell them my caller ID is the corporate number for Maytag and start calling people at 3am with is your refrigerator running that would count as Misleading :) -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --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] Caller ID Spoofing to be banned in the USA
Andrew Joakimsen wrote: The Proposed bill S704 reads It shall be unlawful for any person within the United States, in connection with any telecommunications service or IP-enabled voice service, to cause any caller identification service to transmit misleading or inaccurate caller identification information, Please tell me how you can construe making a call with the the CID of a number in your control to be Misleading or inaccurate You're answering your own question. Forwarding a call with a number that is not the originating number is what (drum roll) a) accurate b) inaccurate If you answered b then see me off-list for your scooby snack. As for possible scenarios, hows this one for size, you change your CID to reflect 1-800-MASTERCARD call around and fish for information: This is John with MASTERCARD services, there's been fraudulent activity on your card and we've suspended it until we can confirm your card number... And that's all she wrote. I've been up and down this road on the VoIP security mailing list: http://lists.virus.org/voipsec-0610/threads.html -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Caller ID Spoofing to be banned in the USA
On 7/3/07, J. Oquendo [EMAIL PROTECTED] wrote: You're answering your own question. Forwarding a call with a number that is not the originating number is what (drum roll) And in a corporate environment, what is the originating number? Is it the main line, the DID, or what? If I am at my house, my calls are routed through my company's Asterisk server with my number at my house showing on my caller ID. What is the originating number? It originated from my house (that's where it all started, because I picked up the phone to dial at my house). What if I select the line that uses my DID as the caller ID? This all gets complicated, and there is not a US Representative or US Senator smart enough to figure this out, that's the scary part. Most probably don't even know what a DID is. By the time it is over with, laws will be passed to outlaw legitimate purposes. However, those manipulating caller ID with illegitimate purposes will continue to do so. Breaking the law is not something those people tend to be concerned about. ___ --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] Caller ID Spoofing to be banned in the USA
Lacy Moore - Aspendora wrote: This all gets complicated, and there is not a US Representative or US Senator smart enough to figure this out, that's the scary part. Most probably don't even know what a DID is. By the time it is over with, laws will be passed to outlaw legitimate purposes. However, those manipulating caller ID with illegitimate purposes will continue to do so. Breaking the law is not something those people tend to be concerned about. I blame the American public who when presented these laws via news reports, sits back like a little puppy waiting to be walked. Its easy to bitch up a storm *after the fact*, but when these issues are thrown around most people are too caught up with moronic issues such as oh noes they done arrested poor little Paris Hilton. Many loathe the ACLU, EPIC, EFF and the things these groups do and its likely because many don't understand why many fight the fights they do. Instead of a what do I do now ... the sky is falling the sky is falling method of taking, people need to really start paying attention to what is going on in government before insanely overbroad and moronic laws are passed. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Caller ID Spoofing to be banned in the USA
Actually, I *NEED* to change the caller ID. Here's why... Someone dials into my DID, their caller ID reflects their (cell, home office, etc...) The call then rings my VoIP phones. It then announces Outside Transfer after 3 rings, at which time it rings my VoIP phones AND my cell phone. If PSTN gateway providers lock the callerid to my DID and I have no way to change it, then I have no idea whom is calling me. And that is a requirement... If I'm sitting in a meeting room with 15 people, I need to know if whomever is calling me is worth the interruption. And frankly, _*NO*_... I don't want to give anyone my cell number. Once you give out the cell number, people call you on it before they attempt any other number. J. Oquendo wrote: Andrew Joakimsen wrote: The Proposed bill S704 reads It shall be unlawful for any person within the United States, in connection with any telecommunications service or IP-enabled voice service, to cause any caller identification service to transmit misleading or inaccurate caller identification information, Please tell me how you can construe making a call with the the CID of a number in your control to be Misleading or inaccurate You're answering your own question. Forwarding a call with a number that is not the originating number is what (drum roll) a) accurate b) inaccurate If you answered b then see me off-list for your scooby snack. As for possible scenarios, hows this one for size, you change your CID to reflect 1-800-MASTERCARD call around and fish for information: This is John with MASTERCARD services, there's been fraudulent activity on your card and we've suspended it until we can confirm your card number... And that's all she wrote. I've been up and down this road on the VoIP security mailing list: http://lists.virus.org/voipsec-0610/threads.html ___ --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 ___ --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] Caller ID Spoofing to be banned in the USA
Many times the news does not carry information about bills before congress. So the only time we hear about them is after the fact. I blame the news in the US as they are the ones initiating the stupid Paris Hilton stories instead of the Real news. Bob R J. Oquendo wrote: Lacy Moore - Aspendora wrote: This all gets complicated, and there is not a US Representative or US Senator smart enough to figure this out, that's the scary part. Most probably don't even know what a DID is. By the time it is over with, laws will be passed to outlaw legitimate purposes. However, those manipulating caller ID with illegitimate purposes will continue to do so. Breaking the law is not something those people tend to be concerned about. I blame the American public who when presented these laws via news reports, sits back like a little puppy waiting to be walked. Its easy to bitch up a storm *after the fact*, but when these issues are thrown around most people are too caught up with moronic issues such as oh noes they done arrested poor little Paris Hilton. Many loathe the ACLU, EPIC, EFF and the things these groups do and its likely because many don't understand why many fight the fights they do. Instead of a what do I do now ... the sky is falling the sky is falling method of taking, people need to really start paying attention to what is going on in government before insanely overbroad and moronic laws are passed. ___ --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 ___ --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] Caller ID Spoofing to be banned in the USA
On 7/3/07, Karl J. Vesterling [EMAIL PROTECTED] wrote: And frankly, *NO*... I don't want to give anyone my cell number. Once you give out the cell number, people call you on it before they attempt any other number. You are absolutely correct. I walk down the hall of our office and see people on their cellphone sitting at their desk. Not because they called someone, but because the other person called them on their cellphone. Makes it difficult trying to save a few dollars. Makes me wonder if maybe we should just dump our phone system altogether and just use cellphones. Looking at our cell usage, there is no way we can use any more minutes than we already do, and we'd save the cost of the PRI. Don't even get me started on all the lights left on... You'd think this place was a US gov't building with all the money being wasted. Getting back to subject, though, one of the things I want to implement is a Follow Me type system so that people call the DID and then our system finds our employee, whether it be at their desk or on their cell phone. I haven't got around to that yet. But hopefully that will help. Problem is, we're in the construction business, too, so a lot of times our people are out of the office. ANd honestly, because I haven't got the follow me implemented, I can't really blame anyone other than myself. ___ --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] Caller ID Spoofing to be banned in the USA
Karl J. Vesterling wrote: Actually, I *NEED* to change the caller ID. Here's why... CID internal and external are two different things. If PSTN gateway providers lock the callerid to my DID and I have no way to change it, then I have no idea whom is calling me. And that is a requirement... If I'm sitting in a meeting room with 15 people, I need to know if whomever is calling me is worth the interruption. And who's problem is it that you can no longer screen your calls? You will see its a call from your company whether or not you decide to answer it leaves little for argument here. Not to rant on you but by boo hoo the law is evil because my indentured servant won't tie my shoes is irrelevant and would never get a second listen in any political arena. Unless of course you lobbied the right people. A better argument might have been I run a victim services agencies in which women who were raped assist callers and we need to protect their privacy so we'd like to be able to have their home number reflect our main number to allude to them being located at a central point. Not some God! Now I have to answer my phone! crybabyish babbling. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Caller ID Spoofing to be banned in the USA
No - I agree completely. I am not advocating that the bill is right or what I want. I am just explaining their logic - which as annoying as it is has some merit. As I read it - as long as you aren't doing anything evil - it doesn't really apply. Note: I am not an attorney nor do I play one on TV. Ron Elvis Stephan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryan Laird Sent: Tuesday, July 03, 2007 3:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID Spoofing to be banned in the USA On Jul 2, 2007, at 11:56 PM, Ron Stephan wrote: Please tell me how you can construe making a call with the the CID of a number in your control to be Misleading or inaccurate Sure - it goes like this - The less scrupulous among us might use a spoofed cid to get people to do something they normally wouldn't. Imagine a spoofed CID of your corporate headquarters and somebody calling your employees saying they were HR and needed to confirm SSN numbers... I will let you fill out the rest of the disaster. Trouble is you don't think as evil as some people do. It annoys the hell out of me too - I would love to spoof my cell CID. I would love to have three or four cells with the same CID (all pointing back to my astericks box). It seems damn near impossible hear in Kalifornia. Ron Elvis Stephan Not to not pick, but I think you went beyond what Andrew was saying... Misleading or inaccurate, I would read this to imply that I'm not miss leading you, I'm not providing inaccurate information I am providing you with a means to contact me back. I'm opening stating who I am and where you can reach me with no malitious attempt. The Misleading or inaccurate part would encompass the scenario you describe above. Much the same the intention isn't to target corporate offices where employee's have DID's but their caller ID shows up as the trunk line which feeds to the building / company operator. Now, if I goto a provider and tell them my caller ID is the corporate number for Maytag and start calling people at 3am with is your refrigerator running that would count as Misleading :) -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations Phone: 703-944-9909 -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --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 __ NOD32 2374 (20070703) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --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] Caller ID Spoofing to be banned in the USA
J. Oquendo wrote: Karl J. Vesterling wrote: Actually, I *NEED* to change the caller ID. Here's why... CID internal and external are two different things. I think Karl was referring to external caller ID. If PSTN gateway providers lock the callerid to my DID and I have no way to change it, then I have no idea whom is calling me. And that is a requirement... If I'm sitting in a meeting room with 15 people, I need to know if whomever is calling me is worth the interruption. And who's problem is it that you can no longer screen your calls? You will see its a call from your company whether or not you decide to answer it leaves little for argument here. Not to rant on you but by boo hoo the law is evil because my indentured servant won't tie my shoes is irrelevant and would never get a second listen in any political arena. Unless of course you lobbied the right people. A better argument might have been I run a victim services agencies in which women who were raped assist callers and we need to protect their privacy so we'd like to be able to have their home number reflect our main number to allude to them being located at a central point. Not some God! Now I have to answer my phone! crybabyish babbling. I ask that you treat people respectfully on the list. The poster has a valid point and does not deserve that kind of response. It's possible to disagree and still be civil, and I've no doubt you're able to do it. Thanks, -Stephen- ___ --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] Caller ID Spoofing to be banned in the USA
Stephen Bosch wrote: I ask that you treat people respectfully on the list. The poster has a valid point and does not deserve that kind of response. It's possible to disagree and still be civil, and I've no doubt you're able to do it. Thanks, Right sorry list for living in a place called reality. Isn't this what discourse is supposed to be? I answer most questions in a matter of factly way. It wasn't directed specifically towards him nor anyone else. Was nothing more than a straighforward realistic point of view. So apologies to anyone who might have felt slighted by comment. Was not my intention regardless of how it sounded. (honestly... I'm just rather blunt) -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Caller ID Spoofing to be banned in the USA
J. Oquendo wrote: Stephen Bosch wrote: I ask that you treat people respectfully on the list. The poster has a valid point and does not deserve that kind of response. It's possible to disagree and still be civil, and I've no doubt you're able to do it. Thanks, Right sorry list for living in a place called reality. In case (as it seems) you're tone deaf about being a dick, there's another example of it right above this sentence. Was not my intention regardless of how it sounded. (honestly... I'm just rather blunt) I read your first couple of posts, and thought, Where does this guy get off? Mr. Know-it-all; rest of us is stupids. Nothing you have said since looks any different. I hope you never have to ask anyone for a favor. You really do seem way too good/smart/clueful for the rest of us mere mortals. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Caller ID Spoofing to be banned in the USA
Looks to me like the law only targets intentionally deceptive spoofing. Dovid B wrote: Anyone know if this is only to bother some one ? I have a client that has a consulting business. The clients call in and his asterisk server call's his cell when he is out of the office. It passes along the CID. I hope the laws don't screw this up for those that change CID on every call for legitimate reasons. - Original Message - *From:* Dean Collins mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Thursday, June 28, 2007 5:43 PM *Subject:* [asterisk-users] Caller ID Spoofing to be banned in the USA Anyone running caller id spoofing applications in the USA running asterisk? Then it’s time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-to-be-outlawed.html Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]+1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --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 ___ --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 ___ --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] Caller ID Spoofing to be banned in the USA
The Proposed bill S704 reads It shall be unlawful for any person within the United States, in connection with any telecommunications service or IP-enabled voice service, to cause any caller identification service to transmit misleading or inaccurate caller identification information, Please tell me how you can construe making a call with the the CID of a number in your control to be Misleading or inaccurate On 7/2/07, Dovid B [EMAIL PROTECTED] wrote: Anyone know if this is only to bother some one ? I have a client that has a consulting business. The clients call in and his asterisk server call's his cell when he is out of the office. It passes along the CID. I hope the laws don't screw this up for those that change CID on every call for legitimate reasons. - Original Message - *From:* Dean Collins [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Thursday, June 28, 2007 5:43 PM *Subject:* [asterisk-users] Caller ID Spoofing to be banned in the USA Anyone running caller id spoofing applications in the USA running asterisk? Then it's time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-to-be-outlawed.html Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -- ___ --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 ___ --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 ___ --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] Caller ID Spoofing to be banned in the USA
Please tell me how you can construe making a call with the the CID of a number in your control to be Misleading or inaccurate Sure - it goes like this - The less scrupulous among us might use a spoofed cid to get people to do something they normally wouldn't. Imagine a spoofed CID of your corporate headquarters and somebody calling your employees saying they were HR and needed to confirm SSN numbers... I will let you fill out the rest of the disaster. Trouble is you don't think as evil as some people do. It annoys the hell out of me too - I would love to spoof my cell CID. I would love to have three or four cells with the same CID (all pointing back to my astericks box). It seems damn near impossible hear in Kalifornia. Ron Elvis Stephan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Monday, July 02, 2007 8:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID Spoofing to be banned in the USA The Proposed bill S704 reads It shall be unlawful for any person within the United States, in connection with any telecommunications service or IP-enabled voice service, to cause any caller identification service to transmit misleading or inaccurate caller identification information, Please tell me how you can construe making a call with the the CID of a number in your control to be Misleading or inaccurate On 7/2/07, Dovid B [EMAIL PROTECTED] wrote: Anyone know if this is only to bother some one ? I have a client that has a consulting business. The clients call in and his asterisk server call's his cell when he is out of the office. It passes along the CID. I hope the laws don't screw this up for those that change CID on every call for legitimate reasons. - Original Message - From: Dean Collins mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com Sent: Thursday, June 28, 2007 5:43 PM Subject: [asterisk-users] Caller ID Spoofing to be banned in the USA Anyone running caller id spoofing applications in the USA running asterisk? Then it's time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-to-be-outlawed.html http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-to-be-outlawed.html Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --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 ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Caller ID Spoofing to be banned in the USA
On 28 Jun 2007, at 17:42, J. Oquendo wrote: Dean Collins wrote: Anyone running caller id spoofing applications in the USA running asterisk? Then it’s time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing- about-to-be-outlawed.html Why it means nothing... You're a carrier doing VoIP... Say a managed carrier. You re-sell trunks. One of those trunks maintains their own PBX. PBX admin decides to spoof out and is using a proxy say in India. Hell make it Tor for that matter. What's to prosecute? Prove it happened from where you say it did - remember the burden is on the prosecution. Now as the carrier (me) first thing I'm going to do is track down which trunk it came from... Then go to that client... So what happens if say the client was legitimately owned and had various proxied addresses committing toll fraud. Analogy... Gun dealer sells a .45 to an authorized gun buyer. Gun owner leaves his gun at home. Someone breaks into his home, cracks his gun safe, uses his gun for a crime, re-enters and places the gun back in the safe. Now its known it wasn't the gun owner because he was witnessed by the court system and recorded say at jury duty... What do you do, prosecute him? For what? Negligence? It would be humorous to see how this plays out. To me its more or less voting time let's sign pretend laws for brownie points The situation here in the UK is that the folks who interconnect to the PSTN have to validate that you own/control the number you are sending via IAX or SIP. We had a problem where an internal id was not getting overwritten with a valid PSTN number, one of our suppliers set a default caller-id and another rejected the calls. The process is annoying, but it works fine, you have to either use callerids of DIDs you have bought from the same ITSP or fax them a telco bill indicating your rights to that number. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] Caller ID Spoofing to be banned in the USA
Well the gun owner will go to jail! Take a look at your local news. Best regards, Al Bochter Bochter Services --- Need to call use our web phone at the link below http://www.bochterservices.com/voip/iaxphone.php?cn=250 --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Business Opportunity Click Below (9 Min Call) http://www.bochterservices.com/voip/iaxphone.php?cn=18003946919 --- Tim Panton wrote: On 28 Jun 2007, at 17:42, J. Oquendo wrote: Dean Collins wrote: Anyone running caller id spoofing applications in the USA running asterisk? Then it’s time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing- about-to-be-outlawed.html Why it means nothing... You're a carrier doing VoIP... Say a managed carrier. You re-sell trunks. One of those trunks maintains their own PBX. PBX admin decides to spoof out and is using a proxy say in India. Hell make it Tor for that matter. What's to prosecute? Prove it happened from where you say it did - remember the burden is on the prosecution. Now as the carrier (me) first thing I'm going to do is track down which trunk it came from... Then go to that client... So what happens if say the client was legitimately owned and had various proxied addresses committing toll fraud. Analogy... Gun dealer sells a .45 to an authorized gun buyer. Gun owner leaves his gun at home. Someone breaks into his home, cracks his gun safe, uses his gun for a crime, re-enters and places the gun back in the safe. Now its known it wasn't the gun owner because he was witnessed by the court system and recorded say at jury duty... What do you do, prosecute him? For what? Negligence? It would be humorous to see how this plays out. To me its more or less voting time let's sign pretend laws for brownie points The situation here in the UK is that the folks who interconnect to the PSTN have to validate that you own/control the number you are sending via IAX or SIP. We had a problem where an internal id was not getting overwritten with a valid PSTN number, one of our suppliers set a default caller-id and another rejected the calls. The process is annoying, but it works fine, you have to either use callerids of DIDs you have bought from the same ITSP or fax them a telco bill indicating your rights to that number. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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 Inbound (clean). Database: 000752-7, 07/01/2007 - 7/1/2007 12:26:58 PM ___ --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] Caller ID Spoofing to be banned in the USA
Al Bochter wrote: Well the gun owner will go to jail! Take a look at your local news. If you own a gun, it's your responsibility to keep it secure. I don't know of an OECD juridiction where that's not the case. -Stephen- ___ --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] Caller ID Spoofing to be banned in the USA
I think you should be able to spoof your caller id to a number you are in control of. Like a toll free number, your main inbound and/or a number that goes to that ext. I think it is a big pain that anyone can spoof your cellular number and if you don't use a password can check your voicemail. How I read the upcoming law that is how it is going to be that you can spoof to a number that you are in control of. And I am fine with that. On the Asterisk server we use I have one inbound trunk that our toll free rings to and 4 outbound trunks that have no caller to them there are not any DID set to them. So for my outbound what would my provider set my caller ID to? Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Stephen Bosch wrote: Al Bochter wrote: Well the gun owner will go to jail! Take a look at your local news. If you own a gun, it's your responsibility to keep it secure. I don't know of an OECD juridiction where that's not the case. -Stephen- ___ --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 Inbound (clean). Database: 000752-7, 07/01/2007 - 7/1/2007 2:16:13 PM ___ --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] Caller ID Spoofing to be banned in the USA
Anyone know if this is only to bother some one ? I have a client that has a consulting business. The clients call in and his asterisk server call's his cell when he is out of the office. It passes along the CID. I hope the laws don't screw this up for those that change CID on every call for legitimate reasons. - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, June 28, 2007 5:43 PM Subject: [asterisk-users] Caller ID Spoofing to be banned in the USA Anyone running caller id spoofing applications in the USA running asterisk? Then it's time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-to-be-outlawed.html Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -- ___ --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___ --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] Caller ID Spoofing to be banned in the USA
Anyone running caller id spoofing applications in the USA running asterisk? Then it's time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-t o-be-outlawed.html Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --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] Caller ID Spoofing to be banned in the USA
Dean Collins wrote: Anyone running caller id spoofing applications in the USA running asterisk? Then it’s time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-to-be-outlawed.html Why it means nothing... You're a carrier doing VoIP... Say a managed carrier. You re-sell trunks. One of those trunks maintains their own PBX. PBX admin decides to spoof out and is using a proxy say in India. Hell make it Tor for that matter. What's to prosecute? Prove it happened from where you say it did - remember the burden is on the prosecution. Now as the carrier (me) first thing I'm going to do is track down which trunk it came from... Then go to that client... So what happens if say the client was legitimately owned and had various proxied addresses committing toll fraud. Analogy... Gun dealer sells a .45 to an authorized gun buyer. Gun owner leaves his gun at home. Someone breaks into his home, cracks his gun safe, uses his gun for a crime, re-enters and places the gun back in the safe. Now its known it wasn't the gun owner because he was witnessed by the court system and recorded say at jury duty... What do you do, prosecute him? For what? Negligence? It would be humorous to see how this plays out. To me its more or less voting time let's sign pretend laws for brownie points -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Caller ID matching
got it. thanx
On 5/24/07, Matthew Yingling [EMAIL PROTECTED] wrote:
We use this macro, which works quite well:
[macro-checkuservoicemail]
; ${ARG1} - Device extension(s) to check for mail
; Usage
; in main context do exten = 1000,1,Macro(checkuservoicemail,101)
exten = s,1,NoOp(Entering CheckUserVoiceMail for ${MACRO_EXTEN})
exten = s,n,GotoIf($[${MACRO_EXTEN} = ${ARG1}]?:NoMatchVM)
exten = s,n,Playback(beep) ; Hack for UIP200 clipping bug
exten = s,n,VoicemailMain([EMAIL PROTECTED] [EMAIL PROTECTED]) ;
Check vmail
exten = s,n,Hangup ; Hangup after checking vmail
exten = s,n(NoMatchVM),NoOp(End checkuservoicemail)
-Original Message-
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] Behalf Of *Mike Hammett
*Sent:* Tuesday, May 22, 2007 9:37 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* RE: [asterisk-users] Caller ID matching
Yeah, I was trying to have it match the caller ID with what they're
dialing so that I don't have a separate entry for every customer.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Rizwan Hisham
*Sent:* Tuesday, May 22, 2007 5:14 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Caller ID matching
I did it anyway. i used another way around to do it:
suppose 88777 is your number
exten= 88777,1,Dial(SIP/you)
exten= 88777/88777,1,VoiceMailMain()
but in this case you will have to make a separate vm extension for every
user.
On 5/22/07, *Rizwan Hisham* [EMAIL PROTECTED] wrote:
well i have tried to solve your problem, making your extensions in my
dialplan and reloading dialplan gives me segmentation fault. im afraid i
cant help u :)
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
On 5/20/07, *Mike Hammett* [EMAIL PROTECTED] wrote:
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own number
to check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing NoOp(IAX2/815748-16, 815748) in new stack
-- Executing Hangup(IAX2/815748-16, ) in new stack
== Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'
May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.
-- Hungup 'IAX2/815748-16'
May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected
connect attempt from 65.182.165.XXX, request '[EMAIL PROTECTED]'
does not exist
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)
exten =
${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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AXVOICE Inc.
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RE: [asterisk-users] Caller ID matching
We use this macro, which works quite well:
[macro-checkuservoicemail]
; ${ARG1} - Device extension(s) to check for mail
; Usage
; in main context do exten = 1000,1,Macro(checkuservoicemail,101)
exten = s,1,NoOp(Entering CheckUserVoiceMail for ${MACRO_EXTEN})
exten = s,n,GotoIf($[${MACRO_EXTEN} = ${ARG1}]?:NoMatchVM)
exten = s,n,Playback(beep) ; Hack for UIP200 clipping bug
exten = s,n,VoicemailMain([EMAIL PROTECTED]) ; Check vmail
exten = s,n,Hangup ; Hangup after checking vmail
exten = s,n(NoMatchVM),NoOp(End checkuservoicemail)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mike Hammett
Sent: Tuesday, May 22, 2007 9:37 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Caller ID matching
Yeah, I was trying to have it match the caller ID with what they're dialing
so that I don't have a separate entry for every customer.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham
Sent: Tuesday, May 22, 2007 5:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID matching
I did it anyway. i used another way around to do it:
suppose 88777 is your number
exten= 88777,1,Dial(SIP/you)
exten= 88777/88777,1,VoiceMailMain()
but in this case you will have to make a separate vm extension for every
user.
On 5/22/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
well i have tried to solve your problem, making your extensions in my
dialplan and reloading dialplan gives me segmentation fault. im afraid i
cant help u :)
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
On 5/20/07, Mike Hammett [EMAIL PROTECTED] wrote:
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own number
to check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing NoOp(IAX2/815748-16, 815748) in new stack
-- Executing Hangup(IAX2/815748-16, ) in new stack
== Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'
May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.
-- Hungup 'IAX2/815748-16'
May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected
connect attempt from 65.182.165.XXX, request '[EMAIL PROTECTED]'
does not exist
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)
exten =
${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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Re: [asterisk-users] Caller ID matching
well i have tried to solve your problem, making your extensions in my
dialplan and reloading dialplan gives me segmentation fault. im afraid i
cant help u :)
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
On 5/20/07, Mike Hammett [EMAIL PROTECTED] wrote:
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own number
to check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing NoOp(IAX2/815748-16, 815748) in new stack
-- Executing Hangup(IAX2/815748-16, ) in new stack
== Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'
May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.
-- Hungup 'IAX2/815748-16'
May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected
connect attempt from 65.182.165.XXX, request '[EMAIL PROTECTED]'
does not exist
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)
exten =
${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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AXVOICE Inc.
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Re: [asterisk-users] Caller ID matching
I did it anyway. i used another way around to do it:
suppose 88777 is your number
exten= 88777,1,Dial(SIP/you)
exten= 88777/88777,1,VoiceMailMain()
but in this case you will have to make a separate vm extension for every
user.
On 5/22/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
well i have tried to solve your problem, making your extensions in my
dialplan and reloading dialplan gives me segmentation fault. im afraid i
cant help u :)
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
On 5/20/07, Mike Hammett [EMAIL PROTECTED] wrote:
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own
number to check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing NoOp(IAX2/815748-16, 815748) in new stack
-- Executing Hangup(IAX2/815748-16, ) in new stack
== Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'
May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.
-- Hungup 'IAX2/815748-16'
May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected
connect attempt from 65.182.165.XXX, request '
[EMAIL PROTECTED]' does not exist
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)
exten =
${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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RE: [asterisk-users] Caller ID matching
Yeah, I was trying to have it match the caller ID with what they're dialing
so that I don't have a separate entry for every customer.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham
Sent: Tuesday, May 22, 2007 5:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID matching
I did it anyway. i used another way around to do it:
suppose 88777 is your number
exten= 88777,1,Dial(SIP/you)
exten= 88777/88777,1,VoiceMailMain()
but in this case you will have to make a separate vm extension for every
user.
On 5/22/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
well i have tried to solve your problem, making your extensions in my
dialplan and reloading dialplan gives me segmentation fault. im afraid i
cant help u :)
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
On 5/20/07, Mike Hammett mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own number to
check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing NoOp(IAX2/815748-16, 815748) in new stack
-- Executing Hangup(IAX2/815748-16, ) in new stack
== Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'
May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.
-- Hungup 'IAX2/815748-16'
May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected connect
attempt from 65.182.165.XXX, request '[EMAIL PROTECTED]' does not
exist
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
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Software Engineer
AXVOICE Inc.
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[asterisk-users] Caller ID matching
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own number to
check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing NoOp(IAX2/815748-16, 815748) in new stack
-- Executing Hangup(IAX2/815748-16, ) in new stack
== Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'
May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.
-- Hungup 'IAX2/815748-16'
May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected connect
attempt from 65.182.165.XXX, request '[EMAIL PROTECTED]' does not
exist
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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[asterisk-users] Caller ID not getting to analog extensions
Hi Folks, Recent installation with a simple TDM11B (one FXO, one FXS) that I've set up (at home). I am receiving callerID fine from the telco, as it shows up in my call detail records, AND on 2 SIP phones. However, I'm not reliably receiving it (that is, very seldom does it come through) on the analog phones. Any ideas on where to check configurations, etc? I haven't encountered this issue before (my other installations are always much larger than this one for home). -- Barry D. Hassler ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID not getting to analog extensions
On Mon, Feb 26, 2007 at 04:31:07PM -0500, Barry D. Hassler wrote: Recent installation with a simple TDM11B (one FXO, one FXS) that I've set up (at home). I am receiving callerID fine from the telco, as it shows up in my call detail records, AND on 2 SIP phones. However, I'm not reliably receiving it (that is, very seldom does it come through) on the analog phones. Any ideas on where to check configurations, etc? I haven't encountered this issue before (my other installations are always much larger than this one for home). Two hipshots: How *many* analog phones on your one FXS? And is it possible that the system is sending CNAME, not just CNID, and the phones don't do names, and are confused? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID not getting to analog extensions
There are 3 or 4 analog phones connected on the FXS port. Only 2 of them have callerID. On the CNAME as opposed to CNID, have NO idea! The callerID worked fine on these phones until I cut them over to the asterisk server this weekend. On 2/26/07, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Feb 26, 2007 at 04:31:07PM -0500, Barry D. Hassler wrote: Recent installation with a simple TDM11B (one FXO, one FXS) that I've set up (at home). I am receiving callerID fine from the telco, as it shows up in my call detail records, AND on 2 SIP phones. However, I'm not reliably receiving it (that is, very seldom does it come through) on the analog phones. Any ideas on where to check configurations, etc? I haven't encountered this issue before (my other installations are always much larger than this one for home). Two hipshots: How *many* analog phones on your one FXS? And is it possible that the system is sending CNAME, not just CNID, and the phones don't do names, and are confused? Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Barry D. Hassler ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id not transferred to SIP device
Am Mittwoch, 10. Januar 2007 22:49 schrieb Yuan LIU: From: Tobias Unsleber [EMAIL PROTECTED] I'm wondering why asterisk is not transferring the callerid to the sip device. Scenario as follows: sangoma --- zaptel --- asterisk --- sip --- SIP-Device zaptel is reporting the callerid, but in the sip packages the sip-address shows unknown as user part, as this sip debug package shows: Have you set up a callerid for your Asterisk box? (Could be anything.) I got Asterisk as caller ID before setting callerid. Afterward (as I recall the sequence of events) I get caller's ID. Hello Yuan, I'm sorry but I don't understand what you mean by saying setup a caller id for the asterisk box itself. In which file I have to set up a caller id for the asterisk box? sip.conf? zapata.conf? dialplan? Regards, Tobias ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id not transferred to SIP device
From: Tobias Unsleber [EMAIL PROTECTED] Am Mittwoch, 10. Januar 2007 22:49 schrieb Yuan LIU: From: Tobias Unsleber [EMAIL PROTECTED] I'm wondering why asterisk is not transferring the callerid to the sip device. Scenario as follows: sangoma --- zaptel --- asterisk --- sip --- SIP-Device zaptel is reporting the callerid, but in the sip packages the sip-address shows unknown as user part, as this sip debug package shows: Have you set up a callerid for your Asterisk box? (Could be anything.) I got Asterisk as caller ID before setting callerid. Afterward (as I recall the sequence of events) I get caller's ID. Hello Yuan, I'm sorry but I don't understand what you mean by saying setup a caller id for the asterisk box itself. In which file I have to set up a caller id for the asterisk box? sip.conf? zapata.conf? dialplan? Regards, Tobias I did in sip.conf, in [general]. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] caller id not transferred to SIP device
Hello, I'm wondering why asterisk is not transferring the callerid to the sip device. Scenario as follows: sangoma --- zaptel --- asterisk --- sip --- SIP-Device zaptel is reporting the callerid, but in the sip packages the sip-address shows unknown as user part, as this sip debug package shows: Executing Dial(Zap/62-1, SIP/123|25|d) in new stack We're at 172.31.253.80 port 10460 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 192.168.11.47:2075: INVITE sip:[EMAIL PROTECTED]:2075;line=gv8x1x75 SIP/2.0 Via: SIP/2.0/UDP 172.31.253.80:5060;branch=z9hG4bK5e96f554;rport From: Unknown sip:[EMAIL PROTECTED];tag=as14f7c144 To: sip:[EMAIL PROTECTED]:2075;line=gv8x1x75 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 08:58:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 265 -- Versions: asterisk 1.2.14 zaptel 1.2.12 linux 2.6.15.7 asterisk addons 1.2.4 SIP-Device: I set CallingPress to allowed also, no effect. I think this is for the outgoing caller id presentation. (?) SIP device config(sip show peer) * Name : 123 Secret : Set MD5Secret: Not set Context : wahlplan_international Subscr.Cont. : Not set Language : de AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: 5 Pickupgroup : 5 Mailbox : 123 VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : Expire : 3010 Insecure : no Nat : RFC3581 ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 192.168.11.47 Port 2069 Defaddr-IP : 0.0.0.0 Port 2069 Def. Username: 123 SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none) Status : Unmonitored Useragent: snom360/6.5.1 Reg. Contact : sip:[EMAIL PROTECTED]:2069;line=h9dxgpnb -- Tobias Unsleber VoIP Consultant focus::voip GmbH http://www.focus-voip.de Hausadresse: Robert-Koch-Strasse 9 D-64331 weiterstadt Postfach 10 01 21 D-64201 Darmstadt Tel.: +49 61 51 / 90 67 - 256 FAX : +49 61 51 / 90 67 - 299 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] caller id not transferred to SIP device
From: Tobias Unsleber [EMAIL PROTECTED] Hello, I'm wondering why asterisk is not transferring the callerid to the sip device. Scenario as follows: sangoma --- zaptel --- asterisk --- sip --- SIP-Device zaptel is reporting the callerid, but in the sip packages the sip-address shows unknown as user part, as this sip debug package shows: Have you set up a callerid for your Asterisk box? (Could be anything.) I got Asterisk as caller ID before setting callerid. Afterward (as I recall the sequence of events) I get caller's ID. Yuan Liu Executing Dial(Zap/62-1, SIP/123|25|d) in new stack We're at 172.31.253.80 port 10460 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP ... -- Tobias Unsleber VoIP Consultant ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller Id problem
Dear List,
My problem is that the incoming Caller Id is not displayed on the local analog
phones (connected to a TDM400 card).
I receive the CID correctly from my telco, but when I place the call to the
internal analog line, the CID is not propagated.
An interesting point: when I try to place a new call to an already bridged line,
I see the second call with the CID on the analog phone. The second call is
placed exactly with the same command/config as the first one.
In the debug log I see (for the second call):
-- Launched AGI Script /usr/share/asterisk/agi-bin/incoming.pl
-- AGI Script Executing Application: (Dial) Options: (Zap/2||otw)
-- CPE supports Call Waiting Caller*ID. Sending '/066332XX'
In other words, the CID is transmitted during a Call Waiting, but not during a
normal call. It looks like Asterisk does not send the CID (or send it too soon /
too late) during the first (normal) call.
Any idea is welcome.
Thanks!
AF.
--
*zapata.conf*
usecallerid=yes
usecallingpres=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
treewaycalling=yes
transfer=yes
useincomingcalleridonzaptransfer=yes
...
context=home
signalling=fxo_ks
channel = 1
context=office
signalling=fxo_ks
channel = 2
context=freebox
signalling=fxs_ks
callerid=asreceived
channel = 3
context=francetelecom
signalling=fxs_ks
callerid=asreceived
channel = 4
*extensions.conf*
exten = s,1,Dial(${HOME},,otw)
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Re: [asterisk-users] Caller Id problem
always include a wait before a dial
give the callerid time to get into * before dialing, it arrives
between the first and second ring, if you have * dial after the first
ring it will not be there yet to pass along
On Jan 9, 2007, at 12:16 PM, Anton Frolov wrote:
Dear List,
My problem is that the incoming Caller Id is not displayed on the
local analog
phones (connected to a TDM400 card).
I receive the CID correctly from my telco, but when I place the
call to the
internal analog line, the CID is not propagated.
An interesting point: when I try to place a new call to an already
bridged line,
I see the second call with the CID on the analog phone. The second
call is
placed exactly with the same command/config as the first one.
In the debug log I see (for the second call):
-- Launched AGI Script /usr/share/asterisk/agi-bin/incoming.pl
-- AGI Script Executing Application: (Dial) Options: (Zap/2||otw)
-- CPE supports Call Waiting Caller*ID. Sending '/066332XX'
In other words, the CID is transmitted during a Call Waiting, but
not during a
normal call. It looks like Asterisk does not send the CID (or send
it too soon /
too late) during the first (normal) call.
Any idea is welcome.
Thanks!
AF.
--
*zapata.conf*
usecallerid=yes
usecallingpres=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
treewaycalling=yes
transfer=yes
useincomingcalleridonzaptransfer=yes
...
context=home
signalling=fxo_ks
channel = 1
context=office
signalling=fxo_ks
channel = 2
context=freebox
signalling=fxs_ks
callerid=asreceived
channel = 3
context=francetelecom
signalling=fxs_ks
callerid=asreceived
channel = 4
*extensions.conf*
exten = s,1,Dial(${HOME},,otw)
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Re: [asterisk-users] Caller Id problem
From: Jerry Jones [EMAIL PROTECTED]
always include a wait before a dial
give the callerid time to get into * before dialing, it arrives between
the first and second ring, if you have * dial after the first ring it will
not be there yet to pass along
Is there a way to count number of rings?
Yuan Liu
On Jan 9, 2007, at 12:16 PM, Anton Frolov wrote:
Dear List,
My problem is that the incoming Caller Id is not displayed on the local
analog
phones (connected to a TDM400 card).
I receive the CID correctly from my telco, but when I place the call to
the
internal analog line, the CID is not propagated.
An interesting point: when I try to place a new call to an already
bridged line,
I see the second call with the CID on the analog phone. The second call
is
placed exactly with the same command/config as the first one.
In the debug log I see (for the second call):
-- Launched AGI Script /usr/share/asterisk/agi-bin/incoming.pl
-- AGI Script Executing Application: (Dial) Options: (Zap/2||otw)
-- CPE supports Call Waiting Caller*ID. Sending '/066332XX'
In other words, the CID is transmitted during a Call Waiting, but not
during a
normal call. It looks like Asterisk does not send the CID (or send it too
soon /
too late) during the first (normal) call.
Any idea is welcome.
Thanks!
AF.
--
*zapata.conf*
usecallerid=yes
usecallingpres=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
treewaycalling=yes
transfer=yes
useincomingcalleridonzaptransfer=yes
...
context=home
signalling=fxo_ks
channel = 1
context=office
signalling=fxo_ks
channel = 2
context=freebox
signalling=fxs_ks
callerid=asreceived
channel = 3
context=francetelecom
signalling=fxs_ks
callerid=asreceived
channel = 4
*extensions.conf*
exten = s,1,Dial(${HOME},,otw)
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Re: [asterisk-users] Caller Id problem
thanks, Jerry but I don't thinks it's a problem, since I correctly get the CID from external line (moreover, I do some lookup of the received number in my LDAP database and making some decisions based on it). So when I call the Dial function, the CID is present in asterisk for sure. AF. Jerry Jones wrote: always include a wait before a dial give the callerid time to get into * before dialing, it arrives between the first and second ring, if you have * dial after the first ring it will not be there yet to pass along ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id ring tones for Asterisk Phone
Most SIP phones handle this functionality by recognizing numbers from speed dial or address book entries in the phone itself. I believe that the PolyCom SIP phones do this (IP430, IP501, IP601, IP650). I hope that this is helpful. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 3, 2007, at 11:15 PM, Jeronimo Romero wrote: I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Thanks in advance!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] caller id ring tones for Asterisk Phone
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Andrewartha Sent: 04 January 2007 05:26 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] caller id ring tones for Asterisk Phone Jeronimo Romero wrote: I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Phones that understand the ALERT_INFO SIP header will do it - I'm using Polycoms, but a quick search of the wiki gives: http://www.voip-info.org/tiki-index.php?page=OptiPoint+600+SIP +-+Distictive+ring+using+ALERT_INFO http://www.voip-info.org/wiki/index.php?page=UIP200 http://www.voip-info.org/wiki/view/Siemens+Gigaset+C450IP http://www.voip-info.org/wiki/view/Polycom+auto-answer+config http://www.cyber-cottage.co.uk/wiki/index.php/Aastra_ring_patterns http://www.voip-info.org/wiki/view/SPA-841 http://www.voip-info.org/wiki/view/Snom+Phones http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP Legacy phones connected to a Citel SIP Handset Gateway also support ALERT_INFO: http://www.voip-info.org/wiki/view/Citel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] caller id ring tones for Asterisk Phone
I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Thanks in advance!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id ring tones for Asterisk Phone
On Wed, Jan 03, 2007 at 11:15:19PM -0500, Jeronimo Romero wrote: I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Thanks in advance!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Many phones can do this. Some have only a limited set of tones that don't vary much. Most phones can do the basics. Some let you have some uploaded wav file ringtones. A smaller number such as the SNOM phones and a few others can actually be given the URL of an audio file as the ringtone, and the phone will download it and play that. I haven't tried it, but it should be possible on the SNOM to: a) Have festival, cepstral or other TTS turn the caller id into an audio file (ideally cached) b) Put that audio file on a local web server c) Set the URL of the audio file as the ring tone. You usually set the ring tone with the SIP Alert-Info header, however various phones use different syntaxes. Do a search on voip-info for terms like ringtone and alert-info for instructions on how to set them. Of course, you can also do things like generate the audio and have your computer, or a nearby computer, play the sound so you get it reading the name or number. Or you could generate your own audio files for the people who call you regularly rather than that trick. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id ring tones for Asterisk Phone
Jeronimo Romero wrote: I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Phones that understand the ALERT_INFO SIP header will do it - I'm using Polycoms, but a quick search of the wiki gives: http://www.voip-info.org/tiki-index.php?page=OptiPoint+600+SIP+-+Distictive+ring+using+ALERT_INFO http://www.voip-info.org/wiki/index.php?page=UIP200 http://www.voip-info.org/wiki/view/Siemens+Gigaset+C450IP http://www.voip-info.org/wiki/view/Polycom+auto-answer+config http://www.cyber-cottage.co.uk/wiki/index.php/Aastra_ring_patterns http://www.voip-info.org/wiki/view/SPA-841 http://www.voip-info.org/wiki/view/Snom+Phones http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] caller ID authentication
Is there a utility or srcipt in asterisk which accepts calls based on caller ID and gives a busy signal if the caller ID is not on the list. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller ID authentication
Vernier Umali wrote: Is there a utility or srcipt in asterisk which accepts calls based on caller ID and gives a busy signal if the caller ID is not on the list. Thanks Search the Wiki or Mailing List archives for the ex-girlfriend option. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller ID authentication
Thanks On 12/13/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Vernier Umali wrote: Is there a utility or srcipt in asterisk which accepts calls based on caller ID and gives a busy signal if the caller ID is not on the list. Thanks Search the Wiki or Mailing List archives for the ex-girlfriend option. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller ID authentication
I looked at the ex-girlfriend option and it's just part of what I needed. What I do want is to setup a whitelist or numbers which can access the asterisk box and its extensions. All other numbers will be given a congestion or busy tone regardless of what extension they are trying to reach. It would be better that the whitelist is in an external database of list that asterisk can look up. On 12/13/06, Vernier Umali [EMAIL PROTECTED] wrote: Thanks On 12/13/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Vernier Umali wrote: Is there a utility or srcipt in asterisk which accepts calls based on caller ID and gives a busy signal if the caller ID is not on the list. Thanks Search the Wiki or Mailing List archives for the ex-girlfriend option. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller ID authentication
Use astdb for such apps. Look at Lookupblacklist, similarly, you can set up ur whitelist http://www.asteriskguru.com/tutorials/lookupblacklist.html Vernier Umali wrote: I looked at the ex-girlfriend option and it's just part of what I needed. What I do want is to setup a whitelist or numbers which can access the asterisk box and its extensions. All other numbers will be given a congestion or busy tone regardless of what extension they are trying to reach. It would be better that the whitelist is in an external database of list that asterisk can look up. On 12/13/06, Vernier Umali [EMAIL PROTECTED] wrote: Thanks On 12/13/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Vernier Umali wrote: Is there a utility or srcipt in asterisk which accepts calls based on caller ID and gives a busy signal if the caller ID is not on the list. Thanks Search the Wiki or Mailing List archives for the ex-girlfriend option. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller ID authentication
Thanks a lot. I think that's what I needed On 12/13/06, Benjamin Jacob [EMAIL PROTECTED] wrote: Use astdb for such apps. Look at Lookupblacklist, similarly, you can set up ur whitelist http://www.asteriskguru.com/tutorials/lookupblacklist.html Vernier Umali wrote: I looked at the ex-girlfriend option and it's just part of what I needed. What I do want is to setup a whitelist or numbers which can access the asterisk box and its extensions. All other numbers will be given a congestion or busy tone regardless of what extension they are trying to reach. It would be better that the whitelist is in an external database of list that asterisk can look up. On 12/13/06, Vernier Umali [EMAIL PROTECTED] wrote: Thanks On 12/13/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Vernier Umali wrote: Is there a utility or srcipt in asterisk which accepts calls based on caller ID and gives a busy signal if the caller ID is not on the list. Thanks Search the Wiki or Mailing List archives for the ex-girlfriend option. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 20:41 + schrieb David Bath:
Hi,
Thanks for quick response.
I changed it as you suggested, but it has the same effect:
In the console I get:
--Executing
Set(SIP/604625-b79140a8,CALLERID(number)=44${CALLERID(number)}) in
new stack
It's running the IF code correctly, but in the true it's just not
evaluating the variable...
Well, perhaps the IF hinders evaluation from happening?
It is by far not as elegant, but you could try
exten=123456,1,GotoIf($[${REGEX(^0..)} = 1]?2:3)
exten=123456,2,Set(CALLERID(num)=44${CALLERID(num):1})
exten=123456,3,ContinueYourDialplanHere
Btw. it should be CALLERID(num), not CALLERID(number), right?
BR
Anselm
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RE: [asterisk-users] Caller ID Rewrite
Hi All,
First, Edwin thanks for the suggestion in the previous email about
Regex. This unfortunately did not work... I believe it was correctly
evaluation the true condition (i.e. I got the same behaviour).
Anselm, thanks! This way does do it. I believe you must be correct -
the variables are not evaluated when they are the true or false part
of an IF function. I wonder if anyone knows if this is a known bug, or
whether it should be perhaps raised?
On the CALLERID(num) vs CALLERID(number) well. There seems to be
quite a lot of conflicting documentation. The upshot is I'm using
CALLERID(number) and CALLERID(name) and they both seem to work fine..
Thanks to all who made suggestions... my nice little rule is working now
:)
Cheers,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: 02 December 2006 09:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 20:41 + schrieb David Bath:
Hi,
Thanks for quick response.
I changed it as you suggested, but it has the same effect:
In the console I get:
--Executing
Set(SIP/604625-b79140a8,CALLERID(number)=44${CALLERID(number)}) in
new stack
It's running the IF code correctly, but in the true it's just not
evaluating the variable...
Well, perhaps the IF hinders evaluation from happening?
It is by far not as elegant, but you could try
exten=123456,1,GotoIf($[${REGEX(^0..)} = 1]?2:3)
exten=123456,2,Set(CALLERID(num)=44${CALLERID(num):1})
exten=123456,3,ContinueYourDialplanHere
Btw. it should be CALLERID(num), not CALLERID(number), right?
BR
Anselm
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[asterisk-users] Caller ID Rewrite
Hi All,
I have a quick query which I'm sure someone will have done before.
Essentially, I have a 3rd party desktop app which does number lookup in
Outook via the manager interface. Works wonderfully. However, it's
not very clever in the number matching. I have all my contacts stored
in +country code number format. My service provider passes all
numbers, apart from UK numbers, to me in this format. Hence, UK number
lookups don't work correctly.
So onto the problem... I'm trying to write a quick on-liner which will
fix up incoming UK format numbers (0 + 10digits) into (44 + 9 digits).
I got as far as this:
exten = incoming extension,1,Set(foo=${IF(REGEX(^0[1-9][0-9]\{9\}$
{CALLERID(number)})?Set(CALLERID(number)=44${CALLERID(number):1})})
The regex is correctly triggered, and the Set(CALLERID(number)= xxx )
method is called, but I am struggling to concatenate the two strings.
I'm trying to set the new callerid to be 44 concatenated with the
original callerid without the leading 0.
Any wisdom would be greatly appreciated.
Cheers,
Dave
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Re: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 14:26 + schrieb David Bath:
So onto the problem… I’m trying to write a quick on-liner which will
fix up incoming UK format numbers (0 + 10digits) into (44 + 9 digits).
I got as far as this:
exten = incoming extension,1,Set(foo=${IF(REGEX(^0[1-9][0-9]\{9
\}$ {CALLERID(number)})?Set(CALLERID(number)=44
${CALLERID(number):1})})
I would try something like
exten = _0[1-9]X.,1,Goto(0044${EXTEN:1},1)
(All numbers beginning null not-null will be rewritten to 0044 plus
the number without the leading zero)
Hth
Anselm
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RE: [asterisk-users] Caller ID Rewrite
Hi Anselm,
Thanks for the help...
I'm slightly confused as to your response.
Wouldn't that look for a /dialled/ number in the format _0number try
and jump to another extension 0044number with priority 1?
If so, that's not what I'm trying to achieve.
I have an external SIP provider, and the extension is a 6 digit number,
e.g. 123456. When calls come in, they are always TO: this 6 digit
number..
Hence, the dialplan has
exten = 123456,1,Goto(sipinternal,myphoneextension,1)
at the moment, all incoming calls are forwarded directly to my
deskphone.
What I'm trying to do is first mangle the incoming caller id (i.e. the
FROM: field) so that all numbers come in countrycode + number.
I've made a bit more progress... and my current diaplan entry looks like
this:
exten=123456,1,Set(${IF(REGEX(^0[1-9][0-9]\{9\}$
{CALLERID(number)})?CALLERID(number)=44${CALLERID(number):1})})
exten = 123456,2,Goto(sipinternal,101,1)
BUT! There's a very odd problem (and I'm sure it's my fault..) the
second callerid function is not being evaluated...
I.e if the true condition is met in the IF statement, the command
should distil down to
Set(CALLERID(number)=44${CALLERID(number):1}
Which it does... but, 44${CALLERID(number):1} appears as a string,
instead of being evaluated!
Any ideas why ??
Cheers,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: 01 December 2006 19:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 14:26 + schrieb David Bath:
So onto the problem... I'm trying to write a quick on-liner which will
fix up incoming UK format numbers (0 + 10digits) into (44 + 9 digits).
I got as far as this:
exten = incoming extension,1,Set(foo=${IF(REGEX(^0[1-9][0-9]\{9
\}$ {CALLERID(number)})?Set(CALLERID(number)=44
${CALLERID(number):1})})
I would try something like
exten = _0[1-9]X.,1,Goto(0044${EXTEN:1},1)
(All numbers beginning null not-null will be rewritten to 0044 plus
the number without the leading zero)
Hth
Anselm
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RE: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 20:03 + schrieb David Bath:
Hi Anselm,
Thanks for the help...
I'm slightly confused as to your response.
Wouldn't that look for a /dialled/ number in the format _0number try
and jump to another extension 0044number with priority 1?
If so, that's not what I'm trying to achieve.
Sorry, my brain is in need for a weekend off work. I obviously
understood your question wrong. My fault.
I've made a bit more progress... and my current diaplan entry looks like
this:
exten=123456,1,Set(${IF(REGEX(^0[1-9][0-9]\{9\}$
{CALLERID(number)})?CALLERID(number)=44${CALLERID(number):1})})
exten = 123456,2,Goto(sipinternal,101,1)
BUT! There's a very odd problem (and I'm sure it's my fault..) the
second callerid function is not being evaluated...
I _think_ the IF is a string evaluation, so the format should be like
SET MYVARIABLE = [IF condition? value1 : value2]
(see http://www.voip-info.org/wiki/index.php?page=Asterisk+func+if )
Try
exten=123456,1,Set(CALLERID(number)=
${IF(REGEX(^0[1-9][0-9]\{9\}$ ${CALLERID(number)})?
44${CALLERID(number):1}:${CALLERID(number)})})
(two linebreaks to be removed)
HTH,
Anselm
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RE: [asterisk-users] Caller ID Rewrite
Hi,
Thanks for quick response.
I changed it as you suggested, but it has the same effect:
In the console I get:
--Executing
Set(SIP/604625-b79140a8,CALLERID(number)=44${CALLERID(number)}) in
new stack
It's running the IF code correctly, but in the true it's just not
evaluating the variable...
Cheers,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: 01 December 2006 20:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 20:03 + schrieb David Bath:
Hi Anselm,
Thanks for the help...
I'm slightly confused as to your response.
Wouldn't that look for a /dialled/ number in the format _0number try
and jump to another extension 0044number with priority 1?
If so, that's not what I'm trying to achieve.
Sorry, my brain is in need for a weekend off work. I obviously
understood your question wrong. My fault.
I've made a bit more progress... and my current diaplan entry looks
like
this:
exten=123456,1,Set(${IF(REGEX(^0[1-9][0-9]\{9\}$
{CALLERID(number)})?CALLERID(number)=44${CALLERID(number):1})})
exten = 123456,2,Goto(sipinternal,101,1)
BUT! There's a very odd problem (and I'm sure it's my fault..) the
second callerid function is not being evaluated...
I _think_ the IF is a string evaluation, so the format should be like
SET MYVARIABLE = [IF condition? value1 : value2]
(see http://www.voip-info.org/wiki/index.php?page=Asterisk+func+if )
Try
exten=123456,1,Set(CALLERID(number)=
${IF(REGEX(^0[1-9][0-9]\{9\}$ ${CALLERID(number)})?
44${CALLERID(number):1}:${CALLERID(number)})})
(two linebreaks to be removed)
HTH,
Anselm
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Re: [asterisk-users] Caller ID Rewrite
David Bath wrote:
Hi,
Thanks for quick response.
I changed it as you suggested, but it has the same effect:
In the console I get:
--Executing
Set(SIP/604625-b79140a8,CALLERID(number)=44${CALLERID(number)}) in
new stack
It's running the IF code correctly, but in the true it's just not
evaluating the variable...
since the REGEX returns 1 if match. try this instead:
exten=123456,1,Set(CALLERID(number)=
${IF($[REGEX(^0[1-9][0-9]\{9\}$ ${CALLERID(number)}) = 1]?
44${CALLERID(number):1}:${CALLERID(number)})})
Cheers,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: 01 December 2006 20:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 20:03 + schrieb David Bath:
Hi Anselm,
Thanks for the help...
I'm slightly confused as to your response.
Wouldn't that look for a /dialled/ number in the format _0number try
and jump to another extension 0044number with priority 1?
If so, that's not what I'm trying to achieve.
Sorry, my brain is in need for a weekend off work. I obviously
understood your question wrong. My fault.
I've made a bit more progress... and my current diaplan entry looks
like
this:
exten=123456,1,Set(${IF(REGEX(^0[1-9][0-9]\{9\}$
{CALLERID(number)})?CALLERID(number)=44${CALLERID(number):1})})
exten = 123456,2,Goto(sipinternal,101,1)
BUT! There's a very odd problem (and I'm sure it's my fault..) the
second callerid function is not being evaluated...
I _think_ the IF is a string evaluation, so the format should be like
SET MYVARIABLE = [IF condition? value1 : value2]
(see http://www.voip-info.org/wiki/index.php?page=Asterisk+func+if )
Try
exten=123456,1,Set(CALLERID(number)=
${IF(REGEX(^0[1-9][0-9]\{9\}$ ${CALLERID(number)})?
44${CALLERID(number):1}:${CALLERID(number)})})
(two linebreaks to be removed)
HTH,
Anselm
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--
Edwin Lam [EMAIL PROTECTED]
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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[asterisk-users] Caller ID issues
I am going to be on site at one of my recent installs tomorrow and I am hoping to fix an issue with the caller id. I would like suggestions for possible problem areas and so I thought I would give as much details as I can. The system has a Sangoma A200D card in it with 4 FXO ports and 2 FXS, The incoming pstn lines are all 3 part of a hunt group and Att has confirmed the settings for caller id on all 3 lines. A call coming in on the third line has caller id, but the other 2 give the following messages in the CLI Nov 21 16:54:09 ERROR[6039] caller id.c: fsk_serie made mylen 0 (-9) Nov 21 16:54:09 WARNING[6039] chan_zap.c: CallerID feed failed: Success Nov 21 16:54:09 WARNING[6039] chan_zap.c: CallerID returned with error on channel 'Zap/1-1' I have tried, Recompiling my Zaptel and Asterisk version, both SVN checkouts of 1.2 Moving the 3rd line to a port on the other FXO module Having Att confirm the settings on all 3 lines. My plan now is to go on site and double check the wiring for flaws and connect a standard phone to the lines and confirm the presence of caller id. aside from that, I have not had this error on any previous installs so I am asking the list for ideas and potential fixes. Thanks in advance for your help. zapata.conf snip [channels] ;channel defaults usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes immediate=no group=1 context=from-pstn ;rxgain=1.0 ;txgain=2.0 signalling=fxs_ks channel = 1-2 -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID issues
Bruce Reeves wrote: Nov 21 16:54:09 ERROR[6039] caller id.c: fsk_serie made mylen 0 (-9) Nov 21 16:54:09 WARNING[6039] chan_zap.c: CallerID feed failed: Success Nov 21 16:54:09 WARNING[6039] chan_zap.c: CallerID returned with error on channel 'Zap/1-1' I have exactly the same problem with 1.2 debian (Ubuntu) package and a TDM400P card (2 FXO + 2 FXS). The only difference is that feed failed: Success is logged as a warning and not as an error. Since I'm just started to use asterisk, I presumed it happens because of a misconfig. AF. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
