similar to: multi step auth?

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. > > >

Hi every all, since a few weeks I came back to asterisk and tried to install version 1.6. The installation went fine so I decided to buy new dids on Voxbone. I have added the sip peers of Voxbone Belgium1 like this in the sip.conf [81.201.82.39] host=dynamic type=friend insecure=very context=your_context canreinvite=no qualify=no deny=0.0.0.0/0.0.0.0 permit=81.201.82.39/255.255.255.255 but

Hi List, I recently signed up with Voxbone to get some International DIDs. I was just about to purchase a DID this morning... but when I went to get it.... voxbone wanted the end user's address information. So I started to put it in... unfortunately... the end-user is in the U.S....but the only options are for a few select cities in GERMANY! I don't understand. Is there some

Dear, Currently, I'm looking for a UPS that can support NUT. I found an APC Smart-UPS 5000VA 230V SRT, but I'm not 100% that is supported. *APC Smart-UPS 5000VA 230V SRT* http://www.apc.com/shop/nl/nl/products/APC-Smart-UPS-SRT-5000VA-230V/P-SRT5KXLI I searched in http://networkupstools.org/stable-hcl.html and I found that you support "*APC Smart*" I would just to confirm

All, I'm in Dallas, TX, US and am looking for inbound-only DID service with 10+ channels on a SIP trunk. Is anyone on this list doing something similar and have any recommendations for a provider? Of course I'll be routing either SIP/IAX to an asterisk server that will be hosted in a Dallas colocation facility. I found VoxBone.com but they only want to deal with customers that can

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

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.

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

> 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

> 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

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]:

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

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"? >

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

Hello, In about one hour we should be chatting with Tim Behrins of Voxbone about their initiative, iNum. I say "should" because he's the scheduled guest, but I haven't heard from him today :) Next week, we'll be "Hacking VoIP" Feel free to top post your answers, it seems to stimulate conversation. /r http://VoipUsersConference.org for the usual data or jump on

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

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 have 2 PBX's, one in each office (say one in New York, one in Boston). I > have mobile users that can show up at either office and connect their soft > phones. > > > > Is there a very simple DUNDI config available which describes how to set > this up? > > Also, can I have the same outbound trunks setup in each office, so that > calls don't have to

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.26. Asterisk 1.4.26 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves a large assortment of issues reported by the community. For a summary of the changes in this release, please see the release summary: