similar to: Calling Card Application

A little bird whispers to me: Don't expect this particular trick to be un-addressed by various legislatures forever. That window is closing, and the bottom of the window looks very much like a guillotine blade - don't have your head in the wrong place. In any case, as has been discussed on -users before (which is where this thread should go, and thus where I'm relegating it)

Reposted to this list: (http://lists.virus.org/voipsec-0610/msg00046.html) > That's exactly the type of thing that needs to be stopped. If Dell outsourcing calls me from India, the CLI must be their number in India not a faked-in number of some office in the US. That to me is exactly the purpose of this proposed law. It is equivalent to the law regarding FAX calls that has been

On Wed, 10 May 2017, J Montoya or A J Stiles wrote: > Presumably your staff carry mobile phones. What about an app that gets > the ID of the cell tower to which it is connected, and passes it and the > SIM number in a HTTP request to a server you control? The problem is that they are supposed to use the 'site landline' to confirm presence -- not their cell phone with the

Hello everybody! I am having the following problem and since I am a beginner in playing with asterisk, i can't solve it: I am trying to integrate my existing H.323 network in real world telephony by ISDN cards. The problem is that i DON'T want to change all e164 numbers in my h.323 network and my ISDN provider doesn't accept those identities (CIDs). So, i have to spoof the outgoing

Can anyone tell me if there is a Calling Card Platform in which I can use in conjuction with Asterisk that can give me Authentication via the caller id of the user. I don't want a PIN based Calling Card system, but the software to be able to recognize the caller ID information and authenticate the users and allow them to call through the asterisk box. Thanks.

How does one do translation for calls that come in from other pbx's where the incoming caller ID is an internal extension number on their pbx? Eg. when I get a call from Free-World-Dial the CID shows up as "429102" which is essentially their internal extension number sans any routing prefix. To dial the number back I need to dial the extension with FWD's routing prefix

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 dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial).

v2: * Rebased onto Linux v4.4-rc2 * vhost: Refuse to assign reserved CIDs * vhost: Refuse guest CID if already in use * vhost: Only accept correctly addressed packets (no spoofing!) * vhost: Support flexible rx/tx descriptor layout * vhost: Add missing total_tx_buf decrement * virtio_transport: Fix total_tx_buf accounting * virtio_transport: Add virtio_transport global mutex to prevent

v2: * Rebased onto Linux v4.4-rc2 * vhost: Refuse to assign reserved CIDs * vhost: Refuse guest CID if already in use * vhost: Only accept correctly addressed packets (no spoofing!) * vhost: Support flexible rx/tx descriptor layout * vhost: Add missing total_tx_buf decrement * virtio_transport: Fix total_tx_buf accounting * virtio_transport: Add virtio_transport global mutex to prevent

On Thu, Jun 27, 2013 at 04:00:03PM +0800, Asias He wrote: > VM sockets vhost transport implementation. This module runs in host > kernel. > > Signed-off-by: Asias He <asias at redhat.com> Has any thought been given to how this affects migration? I don't see any API for an application to move to a different host and reconnect to a running vsock in guest. I think we could

On Thu, Jun 27, 2013 at 04:00:03PM +0800, Asias He wrote: > VM sockets vhost transport implementation. This module runs in host > kernel. > > Signed-off-by: Asias He <asias at redhat.com> Has any thought been given to how this affects migration? I don't see any API for an application to move to a different host and reconnect to a running vsock in guest. I think we could

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Hello Doug, Friday, December 13, 2019, 11:03:37 AM, you wrote: >> This is exactly what I do - “press 1 for a human” >> Works great > I do this as well, but I also do a database lookup to see if the number > is on our speeddial list and if so, pass the call directly on without > the IVR prompts. I do something similar for calls without caller ID, but I was still getting

Hi Guys, The article "They've Got Your number" in the Dec 2004 issue of WIRED magazine mentions Asterisk PBX (on p.100). The article is about phone phreaks hijacking cell phones with Bluetooth technology along with spoofing CID to pull some clandestine hacks on the PSTN. Anyhow, Asterisk is mentioned as the PBX of choice for an outfit: Telephreaks.org out of Florida that has built

Greetings all.. We have an inbound/outbound PRI installed and terminated on a T400P ? Digium Quad T1 card. We?re seeing an odd problem when sending $CALLERIDNUM when calls from the PRI are forwarded back out to the PSTN over the PRI. The $CALLERIDNUM is not being sent out along with the call. It?s sending the phone number of the PRI itself, rather than the $CALLERIDNUM information. Yes, we can

This series is based on v4.4-rc2 and the "virtio: make find_vqs() checkpatch.pl-friendly" patch I recently submitted. v4: * Addressed code review comments from Alex Bennee * MAINTAINERS file entries for new files * Trace events instead of pr_debug() * RST packet is sent when there is no listen socket * Allow guest->host connections again (began discussing netfilter support with

This series is based on v4.4-rc2 and the "virtio: make find_vqs() checkpatch.pl-friendly" patch I recently submitted. v4: * Addressed code review comments from Alex Bennee * MAINTAINERS file entries for new files * Trace events instead of pr_debug() * RST packet is sent when there is no listen socket * Allow guest->host connections again (began discussing netfilter support with

Seems like this is the best idea (challenge-response), a callback. No matter the callerid, you don't know where the caller is. But if you place a call BACK to the callerid, it's going to go to the destination. Then you either need the phone to be answered, or the phone to be answered and and the challenge entered. Adam Goldberg AGP, LLC +1-202-507-9900 -----Original Message-----

It's probably not practical to have them answering the client's telephone! At a lot of sites, incoming calls would be handled by auto attendant, diverted to answering service, etc. --Don -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sebastian Nielsen Sent: Wednesday, May 10, 2017 2:46 PM To:

Regards! During the use of areski a2billing software I'm getting same problem all the time. Actually, after 15 minutes of speaking to someone over calling card, connection brakes. Installation was as smooth as it could be so I don't think I made same kind of a mess in that domain. This is the only problem in the aplication. In the logs everything seems to be fine. I'am sending You