similar to: Digium G100

Hi All, I have installed a number of Digium G100 devices in many countries like South Korea, Japan, Singapore and Australia. I have just installed two in New Zealand and both sites are having a problem with Caller ID. Incoming calls are dropping the first digit 0 from the caller ID. I was previously using DAHDI and a TE121 device which may have been adding the 0, I'm not too sure about

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

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.

> 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. > > >

Hi there, so far I've been unable to get a Giptel G100 phone to operate with Asterisk - I tried both SIP and MGCP. Since I've been playing with quite a number of different SIP (and also MGCP) user devices I am starting to think that it might be the phone's fault and not mine... ;-> Still: Anyone out there has this working with Asterisk? The symptoms are that the G100 won't

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

I'm trying to sort a DATAFRAME by a column "ID" that contains alphanumeric data. Specifically,"ID" contains integers all preceeded by the character "g" as in: g1, g6, g3, g19, g100, g2, g39 I am using the following code: DATAFRAME=DATAFRAME[order(DATAFRAME1$ID),] and was hoping it would sort the dataframe by ID in the following manner g1, g2, g3, g6, g19,

Hi, I am trying to color the data points on my boxplot but I only get the default black. Not sure why. Any ideas? Thank you, Mark Here is my code. > dat.g100 <- dat.sg[100,] # selected gene (row) from a dataframe of 37 columns > d1 <- as.matrix(dat.g100[,1:18]) # subset 1 > d2 <- as.matrix(dat.g100[,19:37]) # subset 2 # boxplots > par(mfrow=c(2,1)) > boxplot(d1,

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

On 6/26/23 5:19 PM, Jeff LaCoursiere wrote: > 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

Hi, We have been using Voxbone for some time for origination, and they now offer E911 services.? We are trying to set this up and having trouble meeting their authentication requirements. I setup a peer as I normally would, with user/pass as they supplied ("lacoursj", "pass"), but my calls are rejected.? Their support is asking that I follow this auth mechanism: 1st step

> 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

On 6/15/20 2:19 PM, Luca Bertoncello wrote: > Am 15.06.2020 um 20:15 schrieb Jeff LaCoursiere: > > Hi Jeff, > >> 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