similar to: Asterisk: BYE is received late

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

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

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

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

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 -

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 provide documentation for Debian/Ubuntu. Well, there is some sort of guide for CentOS around. As

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

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

I am thinking of buying a Logitech C920S Pro HD or C922 webcam for the necessary video conferencing. Is anyone using that with Zoom or perhaps with Jitsi? Do I need drivers? Any issues? Thank you.

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

Dear, i using this scenario. jitsi---> asterisk---->EPABX------> Local Telephone when i am calling from jitsi to no 88 its giving this message and getting busy tone. == Using SIP RTP CoS mark 5 -- Executing [88 at myphones:1] Dial("SIP/sandeep-00000004", "DAHDI/g0/88,20,rt") in new stack -- Called g0/88 [Nov 15 09:53:54] WARNING[3169]: chan_dahdi.c:7536

Hi, Le ven. 10 juil. 2020 à 16:56, Sylvain Boily <sylvain at wazo.io> a écrit : > > It probably can help you: > > https://github.com/wazo-platform/wazo-provd-plugins/blob/master/plugins/xivo-jitsi/1/templates/base.tpl > > Sylvain > > Yes, provided example was exactly what I was after ! Thank you very much ! -------------- next part -------------- An HTML attachment

A most puzzling network conundrum has arisen while I was attempting to create a virtual network behind a virtual router which in turn connects the virtual network to my real network. My machine (192.168.103.23) is on the network with my router (192.168.103.1). The virtual router, tiara, has to connect my 192.168.103.* network with the virtual 10.0.0.* network which comprises two other virtual

Hello, I'm trying to use presence with PJSIP and I have a "issue". I created correctly hint priorities like: exten => 1000,hint,PJSIP/1000 exten => 1001,hint,PJSIP/1001 Extension 1000 can subscribe extension 1001 y vice-versa. The problem is when the extension 1000 make or receive a call. In the softphone where the extension is present on buddy list, the extension appear

I have an iPad Air. Nice kit and good battery life but experience has shown the iPad is just a little too opaque and network dependent for my taste. I was wondering if there existed a similarly sized/robust tablet that can run CentOS-6/7 that anyone would recommend? I am looking from something that I can take on international trips that I can scrub on a regular basis. But, I also desire the

Hi, I created an IAX2 trunk between my old Asterisk 1.4 server (A) and my new one with v. 16.10.0 (B). The trunk seems to be up, and the calls are initiated, eg. an extension from A can dial an extension in B which rings. However, as soon as the extension in B answers, the call is terminated. This is what I see in the console of B: -- Called PJSIP/4053 -- PJSIP/4053-00000002 is ringing

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

Processing Conflict: speexdsp-1.2-0.9.rc3.el6.x86_64 conflicts speex <= 1.2-0.21.rc1 I am loath to replace things on my primary workstation as I have far too much to do as it is without dealing with self-inflicted injuries. However, I do use Jitsi as a softphone and the latest version has a dependency on a package in EPEL which replaces something from the base distro. Can someone inform me

Hello, We've been using Opus for VoIP calls in Jitsi and we're very happy with the sound quality it provides. Thank you for the great work! We are thinking of updating to 1.1-beta and wanted to know if there's any reason not to do so at this point, and wait for the 1.1 release instead. Is the beta expected to be as stable as the 1.0.x series, or are there any known flaws still to