similar to: PMS integration

Hi, A potential client (hotel) has a Property Management System that talks the "Mitel" protocol to their current Mitel PBX in order to receive CDRs (which end up being rated by the PMS system and charged back to guests). Does anyone know of any (free or otherwise) docs on this protocol, or better still have experience interfacing asterisk in a hotel situation like this? The PMS

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.

I'm pretty sure that means your are using a non-color capable terminal, or your termtype variable is incorrect.  What are you using for a terminal emulator? *Jeff LaCoursiere* STRATUSTALK, INC. / CTO Phone: *+1 703.496.4990 x108* Mobile: *+1 815.546.6599* Email: *jeff at stratustalk.com* <mailto:jeff at stratustalk.com> Website: *https://www.stratustalk.com* Address: *One

Hi, I had posted this a few hours ago, but got caught in moderation for size.  I trimmed down the pic and attached. I am on an Ubuntu 16 workstation, in an Ubuntu terminal window, ssh'ed to the PBX (amazon instance).  You can see my term type matches yours. I really don't know why yours doesn't work.  Perhaps you can tell us what your terminal emulator is, what you are running it

Hi Luca, On 6/23/20 8:02 AM, Luca Bertoncello wrote: > > I have problem calling someone outside my networks and I have problem > if the peers are in different networks... I may have missed this originally - are you saying you have trouble when internal phones call each other, if they are on different VLAN's?  That's a pretty big deal. I didn't see my post with the graphs

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

Hello Luca, We are still playing with visualization of your data, but I didn't want you to wait any longer for some results.  I think I blame both DT and the Pi :) First, a look at the phone side of your Banana Pi.  The first thing we noticed is there were a LOT more packets in one direction (north towards DT) than the other (towards the phone): jeff at

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

Does anyone know anything about the Fidelio and Mitel protocol for hotel / motel? Are these industry standards or proprietary formats? Are there open standards for communication with Hotel management software's that could be used in conjunction with a custom asterisk deployment? Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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 *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. > > >

I need to get hooked up with this class, I could have students doing projects for homework :) Interested in RTCP? j On 6/26/23 7:45 PM, TTT wrote: > > 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,

Our company can offer VoIP to premises and domestic users and bill the premises as a whole. We need something to enable the hotel owner to bill each guest in a hotel in real time. What solutions do exist presently? (PS: Our radius (and every telephony equipment outside the hotel) does not recognise which room in the hotel initiated the international (VoIP) call, so that's the main problem

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 a Mitel SX2000 with no voicemail. I'm wondering if it would be possible to use Asterisk to meet the need. This is a hotel property with mostly analog extensions. Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050628/25fd0ee3/attachment.htm

If you just want something easy to use out of the box, install the FreePBX distro. Cheers, j On 4/6/23 11:08 AM, Steve Matzura wrote: > Anthony, > > > No, I had no intention of doing any of those things, for I know not > what they are or why I would need or want to be doing them. Maybe I > should have just stuck with the original idea of installing from > Debian distro.

Hi folks, I've been trying to evaluate T.38 termination providers for a document fax service we currently use (really!) US Robotics Courier modems for, on POTs lines. I managed to make it work through one of our existing upstreams, and reliably, but sadly they are the one upstream I was planning to cancel shortly. Of our other five upstreams, including a new one that I signed up with

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

I have two issues... First I am working with a hotel software vendor to include an automated way to turn vmail on and off while clearing it at the same time. The vendor is looking to interface via serial cable as they currently do with Mitel systems. i am willling to work with them on an IP interface but I am not so sure on how to implement it in asterisk. Does anyone know of a way that may be