similar to: Stir-Shaken for asterisk

> > 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 >

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.

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 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

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

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

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

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

Am 15.06.2020 um 21:50 schrieb Luca Bertoncello: > What do you mean now? If I can use the full available band or if I can > download exactly 50Mbs? > The answer to the first question is: YES! That's why I use a traffic > shaper... ;) > The answer to the second question is: NO. I made a speedtest right now > and I get only ~18Mbps download. And some other information, too.

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 2020-05-31 18:39, Ira wrote: > > I typed this at the terminal prompt: export ASTERISK_PROMPT="%C31[%H]: " > > Typing at the same place: echo $TERM returns xterm > > And now I have colored prompts at the Asterisk command line, so I can > assure you it can work. Kind of cool, 14 years using Asterisk and > because of your question, I now have colored

> On 2020-05-31 15:59, Antony Stone wrote: > On Sunday 31 May 2020 at 15:44:46, Fourhundred Thecat wrote: > > "%Cn[;n] - Change terminal foreground (and optional background) color to > specified A full list of colors may be found in include/asterisk/term.h" > > So, try: > > export ASTERISK_PROMPT="%C31[%H]: " > > (I got 31 from reading the

Hi, We are working on a product to analyze pcap files of VoIP calls. So far it does a reasonable job of analyzing the frequency distribution of packets in both directions, pointing out which direction packet loss / bad jitter occurs.  If you can trap the traffic on the outside and the inside of your Banana Pi and send me the pcap files, I would be happy to run it through our analyzer as

I *am* doing that, as I assumed it would be required just for the 911 mapping we have provided, but that doesn't change the SIP header. Cheers, j On 05/08/2018 02:41 PM, Khalil Khamlichi wrote: > try setting the callerid with > > same => n,Set(CALLERID(all)=17864089672 <17864089672>) > > ofcourse for each customer you will need to provide his own did. > > >

Am 23.06.2020 14:49, schrieb Marek Greško: Hi Marek, > this could be ip address of the different interface on the same box. I > think it works like expected. The only exception would be if the sip > peer ignores the icmp packet unreachable. But I doubt this is the Do you mean "my Linux-Box ignores ICMP packet unreachable" or "Deutsche Telekom ignores them"? >

Hello, how can I change the color of the asterisk prompt to red ? I read in the wiki that I can use %Cn[;n] https://wiki.asterisk.org/wiki/display/AST/Asterisk+CLI+Configuration But what does this mean ? There is no example how to actually use it. where do I put it? What syntax is that anyway? How do I specify red ? I currently have this in my environment: export ASTERISK_PROMPT="[%H]:

On 6/26/23 9:00 AM, Joshua C. Colp wrote: > On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: > > I am connecting to the ARI with subscribe all, so I can see > channels being created. I now want to extract a variety of header > variables (at the moment the from and to tag).  I tried to read > them from the ARI but Asterisk refuses since the

I’m in training, so I have to demonstrate something SIP related. I figure it would be cool to hack a call, hanging it up while in progress from outside Asterisk. Doing so will demonstrate use/knowledge of ARI, AMI, SIP, route-sets, UDP, etc. Practical value: zero :) Who knows, maybe this will have an actual application for someone someday. In practical terms I think building a proxy

Am 14.06.2020 um 17:05 schrieb Antony Stone: Hi Antony, > You mean that the Thomson phone is registering to Deutsche Telekom? > > I thought it was registering to your Asterisk server. Sorry, I didn't read correctly your test 2b... Normally my Thomson phone is registering to my Asterisk server. I tried to register the Thomson phone directly to Telekom's server, to check if the

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