similar to: Adding SIP method MESSAGE to Allow header

Hi All, I have configured WebRTC according to the install document. The clients register correctly. I'm use SIPjs. The clients are able to send messages to the server. The SIP debug shows the messages being received. However I'm stumped for directions on how to route the messages between the clients. Asterisk 11.11.0 Here is my client sip config: [1060] type=friend username=1060 ; The

I am trying to send SMS from my grandstream GXV3240 Asterisk receives the message in a NOTIFY block. How can I get asterisk to run dialplan code when receiving these Notify SMS Message Blocks. I can then route them to my SMS provider. Any ideas are appreciated. Below is debug of a message sent from the phone when received no dialplan code is triggered. I am wounding if I need to

Hi, I'm having trouble configuring Asterisk to respond to an incoming out of call SIP MESSAGE. The transport protocol is TLS and the Asterisk version is 10 (it's old, but I'm kind of stuck with it at the moment). Currently I have roughly the following configuration and handling: sip.conf: [general] accet_outofcall_messages=yes outofcall_message_context=sip-im and extensions.conf

On Sun, May 3, 2020, at 10:33 AM, Jason Pyeron wrote: > > -----Original Message----- > > From: Behalf Of Nicolas Kovacs > > Sent: Sunday, May 3, 2020 2:54 AM > > > > Le 03/05/2020 ? 08:44, Benson Muite a ?crit : > > > They have rpms: > > > > > > https://download.jitsi.org/jitsi/rpm/ > > > > That's the Jitsi desktop

On 05/05/2020 04:43 AM, Erick Perez - Quadrian Enterprises wrote: > Benson, no SELINUX was not enabled. The instance was selected without it > just to make things easier. > I do not have a pull request for the installation manual yet. > > On Tue, May 5, 2020 at 1:21 AM Benson Muite <benson_muite at emailplus.org> > wrote: > >> On Mon, May 4, 2020, at 10:38 PM,

On May 3, 2020 3:33:29 AM EDT, Jason Pyeron <jpyeron at pdinc.us> wrote: >> -----Original Message----- >> From: Behalf Of Nicolas Kovacs >> Sent: Sunday, May 3, 2020 2:54 AM >> >> Le 03/05/2020 ? 08:44, Benson Muite a ?crit : >> > They have rpms: >> > >> > https://download.jitsi.org/jitsi/rpm/ >> >> That's the Jitsi

> -----Original Message----- > From: Behalf Of Nicolas Kovacs > Sent: Sunday, May 3, 2020 2:54 AM > > Le 03/05/2020 ? 08:44, Benson Muite a ?crit : > > They have rpms: > > > > https://download.jitsi.org/jitsi/rpm/ > > That's the Jitsi desktop application. > > > > > and also a scalable installation: > > > >

Le 03/05/2020 ? 08:44, Benson Muite a ?crit?: > They have rpms: > > https://download.jitsi.org/jitsi/rpm/ That's the Jitsi desktop application. > > and also a scalable installation: > > https://github.com/jitsi/jitsi-meet/blob/master/doc/scalable-installation.md That's Debian-specific. > > Which one do you need? None. I need a comprehensive

On Mon, May 4, 2020, at 10:38 PM, Erick Perez - Quadrian Enterprises wrote: > Hi Centos friends. > I had some time to write a spartan tutorial on running the latest stable > Jitsi Video Bridge and Jitsi Meet and Centos 7.7. > I wrote it while testing it so this WORKS and I am currently using it for > fun with the kids. > > I do have the server currently running but blocked by

Benson, no SELINUX was not enabled. The instance was selected without it just to make things easier. I do not have a pull request for the installation manual yet. On Tue, May 5, 2020 at 1:21 AM Benson Muite <benson_muite at emailplus.org> wrote: > > On Mon, May 4, 2020, at 10:38 PM, Erick Perez - Quadrian Enterprises wrote: > > Hi Centos friends. > > I had some time to

HI ALL got small question i use call-limit=1 on peers but call limit is not working if user is not registered on PBX and making calls so the main question is -- how to Disallow CALLS without registering on PBX -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy

On Sun, May 3, 2020, at 9:07 AM, Nicolas Kovacs wrote: > Hi, > > Anyone here managed to run Jitsi Meet on a CentOS 7 server ? As far as I can > tell, it's one of the nicer open source videoconferencing solutions out there. > I gave it a spin on https://suricate.tv and now I'd like to install it on one > of my servers. > > Unfortunately the Jitsi developers only

Hi Centos friends. I had some time to write a spartan tutorial on running the latest stable Jitsi Video Bridge and Jitsi Meet and Centos 7.7. I wrote it while testing it so this WORKS and I am currently using it for fun with the kids. I do have the server currently running but blocked by my firewall. I am willing to allow a few of the people such a Kovacs and others to connect to my Jitsi server

Sorry for the delay here. For some reason the mail from Joshua Colp failed to deliver to my mailbox. So, anyway, I've set up a local scenario on my computer a PJSIP client and Asterisk 11.17.1 (On a fedora linux workstation) with the settings listed below. In this scenario I've used UDP, but I want a configuration that can be used with any transport protocol. I can see that the context

I posted this to the Jitsi users mailing list on Thursday last and have received no replies. Does anyone here have any idea of what is causing this and how to fix it? One other bit of information. When I first installed Jitsi it initially worked for a non-privileged user as I ran it from the gnome menu of my login account. However, at some point it seemed get 'confused' and hung.

Hello, On 2020-07-10 10:39 a.m., Olivier wrote: > Hello, > > 1. I'm looking for an (anomized) example of a Jitsi Desktop > provisioning file compliant with Asterisk  ? > Jitsi Doc mentions it should adhere to Java properties file syntax > (see [1]) but a working example would help. > > If this example file included the following settings, it would be perfect: > -

Hello, 1. I'm looking for an (anomized) example of a Jitsi Desktop provisioning file compliant with Asterisk ? Jitsi Doc mentions it should adhere to Java properties file syntax (see [1]) but a working example would help. If this example file included the following settings, it would be perfect: - sip username, secret, authname, ... - codec G711, RFC2833 - BLF - local or remote phonebook -

Hello. Is there an analog option "outofcall_message_context" for pjsip? or: how to determine that the "call" is an outbound text message? Dmitriy Serov.

I've now replicated my setup on a host with a single IPv4 address and I am still having trouble with the ICE negotiation. I am trying to call from Jitsi to Asterisk through a Prosody XMPP server. Asterisk successfully registers with the XMPP server and appears to be available in the buddy list in Jitsi. Jitsi is being run with the "-4" command line option to use IPv4 only just in

Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI> show hints -= Registered Asterisk Dial Plan Hints =- 30@default : State:Unavailable Watchers 3 29@default :