similar to: Stir Shaken is upon us

On 13.07.20 at 00:17 Joshua C. Colp wrote: > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins <dan at nimblea.pe> wrote: > >> Asterisk 18 will have support based on this asterisk update Matt F did for >> CommCon's sponsor slots >> >> https://youtu.be/eas1csaX-wc >> >> > As well support will go into Asterisk 16 and 17 as well. It's just been

On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins <dan at nimblea.pe> wrote: > Asterisk 18 will have support based on this asterisk update Matt F did for > CommCon's sponsor slots > > https://youtu.be/eas1csaX-wc > > As well support will go into Asterisk 16 and 17 as well. It's just been under active development so we've been waiting for that to finish before

On Sun, Jul 12, 2020 at 11:37 PM Michael Maier <m1278468 at mailbox.org> wrote: > On 13.07.20 at 00:17 Joshua C. Colp wrote: > > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins <dan at nimblea.pe> wrote: > > > >> Asterisk 18 will have support based on this asterisk update Matt F did > for > >> CommCon's sponsor slots > >> > >>

WORLDWIDE EMERGENCY The code below needs to be executed before any SIP or PJSIP call destined to the US network, or soon no call will terminate. This is called Stir-Shaken, a new law from the FCC. If this is not working the whole Asterisk industry will crash, vanish, be gone. I am assuming that the caller ID and the Destination Number are in the variables "${CALLERID(num):-10}"

On Sun, 12 Jul 2020, Saint Michael wrote: > WORLDWIDE EMERGENCY Again? > The code below needs to be executed before any SIP or PJSIP call > destined to the US network, or soon no call will terminate. This is > called Stir-Shaken, a new law from the FCC. If this is not working the > whole Asterisk industry will crash, vanish, be gone. Seen any little chickens lately?

A few weeks... like in a year and a few weeks: https://transnexus.com/blog/2020/fcc-mandates-stir-shaken/ Some interesting bits in there as well, like: "These rules do not apply to providers that lack control of the network infrastructure necessary to implement STIR/SHAKEN." See also: https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN *Jeff LaCoursiere* STRATUSTALK, INC.

Check out Twilio From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Federico Sent: Thursday, August 17, 2023 11:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users at lists.digium.com> Subject: [asterisk-users] Question about Sip Trunks who support Stir Shaken I am looking for a decent provider of SIP

I am looking for a decent provider of SIP Trunks but it has to pass the Stir Shaken token to the next carrier. Does anybody know about any? Sipstation from Sangoma, does not support Stir Shaken. ( Case #01466843 / 0013000000G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ]) Although it's mandatory, somehow they think it's ok. Go figure. -------------- next part -------------- An

Telnyx, 382com, voicetel and as others mentioned BandWidth. I have contacts at 382 and voicetel if you want an intro. On Thu, Aug 17, 2023 at 11:50 PM Federico <federico at digitalipvoice.com> wrote: > I am looking for a decent provider of SIP Trunks but it has to pass the > Stir Shaken token to the next carrier. Does anybody know about any? > Sipstation from Sangoma, does not

On Mon, Jul 13, 2020 at 2:34 PM Saint Michael <venefax at gmail.com> wrote: >> >> There is a big confusion here about Stir Shaken. It is NOT a provider issue. Un fact, all providers are whasing their hands and modifying their swihtches to pass-through the Signature. They cannot sign the call because then the become the responsible party for the call before the FCC, and liable for

On 7/13/20 2:32 PM, Saint Michael wrote: > > There is a big confusion here about Stir Shaken. It is NOT a > provider issue. Un fact, all providers are whasing their hands and > modifying their swihtches to pass-through the Signature. They > cannot sign the call because then the become the responsible party > for the call before the FCC, and liable for any

From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Saint Michael Sent: Tuesday, July 14, 2020 2:35 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Stir Shaken I need to point out the this is factually misleading and materially false: "I think this, being the basis of your whole argument, is the fallacy. S/S is forcing people to take

https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN The Wiki above is misleading in what Stir-Shaken means and how it works. End users cannot get a certificate, they cannot self-certify their calls. Somebody completely misunderstood the model. I am afraid the moment will come and thousands of Asterisk operators will be unable to terminate calls. To start with, the model is a hierarchical

On 13.07.20 at 10:54 Joshua C. Colp wrote: > On Sun, Jul 12, 2020 at 11:37 PM Michael Maier <m1278468 at mailbox.org> wrote: >> One more question, >> what about the pjsip pcap support? Will it be backported to Asterisk 16, >> too? Would be absolutely cool! Debugging encrypted SIP is really a pain. >> > > It can't be backported ... because it already is.

On Mon, 13 Jul 2020 15:44:12 -0400, Matthew Fredrickson wrote: > > On Mon, Jul 13, 2020 at 2:34 PM Saint Michael <venefax at gmail.com> wrote: > >> > >> There is a big confusion here about Stir Shaken. It is NOT a provider issue. Un fact, all providers are whasing their hands and modifying their swihtches to pass-through the Signature. They cannot sign the call

I need to point out the this is factually misleading and materially false: "I think this, being the basis of your whole argument, is the fallacy. S/S is forcing people to take responsibility, for sure, but carriers won't just let their customers leave because they don't want to sign calls. It will force them to make sure they know who their customers are, and make it impossible for

> > There is a big confusion here about Stir Shaken. It is NOT a provider > issue. Un fact, all providers are whasing their hands and modifying their > swihtches to pass-through the Signature. They cannot sign the call because > then the become the responsible party for the call before the FCC, and > liable for any illegal call. Every owner of a PBX that sends calls to the >

In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law. We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are the only ones who don't need to receive the calls in our servers to sign them. We do this over a MySQL call, easily connectable to Asterisk via

> > My company is one if the six service providers approved. We are not ready > yet, probbably next week, since the certificate needs to be issued by the > Certification Authority. As I said, we are the ONLY provider that you may > use with Asterisk remotely, via UnixODBC. The rest of the other providers > will force you to send a call to them. Here is some material for you to

Hello, I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only problem until now which remained was that if dtls_rekey was set to the value other than 0, call hanged up when using chrome after the time where dtls_rekey was set. I suppose that "bad media description" shown in Chrome's window which causes call to fail, has appeared with Chromes newer versions