search for: privaci

On 16 February 2017 at 11:46, James Hogarth <james.hogarth at gmail.com> wrote: > On 16 February 2017 at 11:35, Alice Wonder <alice at domblogger.net> wrote: >> On 02/16/2017 03:28 AM, James Hogarth wrote: >>> >>> On 16 February 2017 at 10:42, Alice Wonder <alice at domblogger.net> wrote: >>>> >>>> On 02/16/2017 02:32 AM, James

Hi all, I’m trying to rewrite Diversion header when call forwarding is done on the phone. The phone sends "302 Moved Temporarily" response and sets Diversion header to a local number, but before Asterisk sends this call towards TSP provider I need to change Diversion header to a full PSTN number. I am using PJSIP_HEADER in a pre-dial handler (configuration is below). On the same

Previous threads have discussed the now deprecated NONE cipher and the rationale for its removal. Recent posts on FCC compliance indicate the usefulness of authentication without privacy for packet radio. I also find this useful for transfers between IPsec protected endpoints. Authentication is then useful for user administration/privileges but encryption is not necessary given the IPsec

Hello all, I would like to use Asterisk to add/modify SIP headers in the INVITE message, to include Privacy information, if the INVITE includes a *67 prefix (or another predefined prefix). That's an example of the INVITE I get: /INVITE sip:*6700112233445 at 192.168.1.100 SIP/2.0 From: "123456789"<sip:*123456789*@192.168.1.100>;tag=333333333 To: <sip:*6700112233445 at

Hi All, I got this figured out, when the privacy is ON at the other end of the server and when we get the Invite message to the server connected to PRI's, just take the details from the invite message in the Dial plan and send the calls as anonymous: exten => _1NXXXXXXXXX,n,Set(PRIVACY=${SIP_HEADER(Privacy)}) exten => _1NXXXXXXXXX,n,ExecIf($["${PRIVACY}" =

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

On 02/16/2017 04:20 AM, James Hogarth wrote: > On 16 February 2017 at 12:02, James Hogarth <james.hogarth at gmail.com> wrote: >> On 16 February 2017 at 11:46, James Hogarth <james.hogarth at gmail.com> wrote: >>> On 16 February 2017 at 11:35, Alice Wonder <alice at domblogger.net> wrote: >>>> On 02/16/2017 03:28 AM, James Hogarth wrote:

Hi folks, My v1.6 Asterisk system logs all Call Detail Records to a PostgreSQL database, including those handled by the Privacy Manager. Unfortunately, even though I can use the CLI to see the information being submitted by anonymous callers to satisfy the demands of the the Privacy Manager, that information is not recorded in the database. Instead, all that is written to it: clid:

On 02/16/2017 03:28 AM, James Hogarth wrote: > On 16 February 2017 at 10:42, Alice Wonder <alice at domblogger.net> wrote: >> On 02/16/2017 02:32 AM, James Hogarth wrote: >>> >>> On 16 February 2017 at 10:17, Alice Wonder <alice at domblogger.net> wrote: >>>> >>>> On 02/16/2017 02:03 AM, James Hogarth wrote: >>>>>

On 16 February 2017 at 12:02, James Hogarth <james.hogarth at gmail.com> wrote: > On 16 February 2017 at 11:46, James Hogarth <james.hogarth at gmail.com> wrote: >> On 16 February 2017 at 11:35, Alice Wonder <alice at domblogger.net> wrote: >>> On 02/16/2017 03:28 AM, James Hogarth wrote: >>>> >>>> On 16 February 2017 at 10:42, Alice

I am attempting to write in incoming context for calls. 1. If the caller id is given and it is not black listed it will Playback a greeting and then right the phone or go to voicemail under busy or unavailable conditions 2. If no caller id is given, then Privacy Manager will ask for the number. I am testing 6145551212 to see if the black list will work 3. If a caller id is given, and it is

"Setting Windows version to winxp Executing early_wine regedit c:\winetrickstmp\set-winver.reg Install of msi2 done You gave the --optin option, so reporting 'msi2' to the winetricks maintainer so he knows which winetricks verbs get used and which don't. Use --optout to disable future reports. winetricks done. " Not sure what this is supposed to be. Is it saying that the

Privacy Guard Glue (PGG) Snapshot 19990628 Released ===================================================== PGG (Privacy Guard Glue) is a well-designed library written in C to add GnuPG support to applications. PGG uses GnuPG (Gnu Privacy Guard) with its coprocessing interface as the workhorse. PGG is free software and includes the complete source code. It is protected under the Gnu

I am having issues with Privacy Manager and Zapateller. If I set callerid="" on a sip user zapateller sends the tones If I set callerid="Anonymous" <8475551212> zapateller doesn't send the tones If I call from a phone after dialing *67 zapateller doesn't send the tones In the last 2 cases, the display on the phone shows -Blocked Call- PrivacyManager always gives

When a user calling over the PSTN network calls one of our SIP users with a restricted number (CLIR), our PSTN gateway is sending us incoming calls with the following additional headers: Proxy-Require: privacy Anonymity: uri Remote-Party-ID: <sip:41781234567@192.168.0.100:4000>;privacy=uri as opposed to Remote-Party-ID: <sip:41781234567@192.168.0.100:4000>;privacy=off when CLIP

Is there a way to use privacy manager without requiring the user to enter their name? Essentially I am just looking for a way to force the called user to enter "1" to accept the call. I don't need a name recording. I want a call to come in, a message to be played, music on hold, call out to the called party, then enter "1" to accept, "2" to reject. Peder

On Tue, Feb 18, 2020 at 2:55 AM David Chisnall via llvm-dev < llvm-dev at lists.llvm.org> wrote: > Hi Tanya, > > On 18/02/2020 00:56, Tanya Lattner wrote: > > I understand not everyone wants to use Google docs which was why I gave > an option to reply via email as well. My intention on using Google docs is > really just based on my experience using it for docs that span

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

On Mon, March 23, 2015 05:24, Nux! wrote: > I find this very, very sad. > I find it unsavoury. We are recommending that acknowledged newbies subscribe to a service known for repeatedly and persistently violating its members' privacy? -- *** E-Mail is NOT a SECURE channel *** James B. Byrne mailto:ByrneJB at Harte-Lyne.ca Harte & Lyne Limited

This is probably a very basic question, but what is the proper controller & routing setup if I''m setting up a RoR site with static pages, like: http://mysite.com/about http://mysite.com/services http://mysite.com/contact I want each of these pages to use a single template file that I can push content to. It seems like overkill to create individual controllers for about,