similar to: Detected alarm on channel 5: Red Alarm

Hi, I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and dahdi driver. Scenario is jitsi-----> asterisk server-----> analog PBX ----> landline phone I configured this scenario as follow in chan_dahdi.conf file ; General options [channels] usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes

I'm trying to integrate my Asterisk box with my call manager 8 server. I can call from the call manager to a phone on asterisk, but I can't call from a phone on asterisk to call manager. Any help would be greatly appreciated. sip.conf [2000] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes [2001] type=friend secret= dtmfmode=rfc2833

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI>

Dear, I have redhat enterprise linux 6.3. after "uname -a" i am getting Linux genesys-dell 2.6.32-279.el6.x86_64 #1 SMP Wed Jun 13 18:24:36 EDT 2012 x86_64 x86_64 x86_64 GNU/Linux now when i am trying to insall dahdi driver on my server i am getting below error. [root at genesys-dell dahdi-linux-complete-2.6.2+2.6.2]# make all make -C linux all make[1]: Entering directory

Hello List: Am a newbie planning to setup Asterisk for the first time. Planning to use PIII 550 Mhz with 256 or 512 MB Ram & 80 GB HDD. Please advise if the above configuration is ok for the application below...if not, please tell me what I need to have a comfortable system. I plan to install minimum CentOS 4.2 Server required with iptables, Samhain & Asterisk & associated programs.

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I've this infrastructure: |voip services| -- |*| -- |cme| -- |isdn| the voip services are logged on my *, then forwarded to number 601 on cme. The isdn calls too are forwarded to 601. On cme I've a timeout X for call-forward noan (no answer) to a specific number on * (5901) that is my x-lite software client. If 5901 is

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, 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

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

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

I have a SIP trunk built between a Cisco CallManager version 8. I can dial the phones registered to the Asterisk PBX from a phone registered to the Call Manager. I've tried to keep the config as small as possible to help the troubleshooting process. Attached is he most recent debug. My Callmanager IP address is 10.169.169.250, Asterisk server is 10.169.169.251 SIP.CONF [6001] type=friend

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 -

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! I have an analog line connected to my asterisk and when I try to answer a call I get this -- Starting simple switch on 'DAHDI/7-1' -- Executing [s at from-pstn:1] Answer("DAHDI/7-1", "") in new stack -- Executing [s at from-pstn:2] Playback("DAHDI/7-1", "vm-intro") in new stack -- <DAHDI/7-1> Playing 'vm-intro' (language

Hi everyone! I have a Digium Card TDM410 But, it appear for me this massege chan_dahdi.c:8061 handle_alarms: Detected alarm on channel 3: Red Alarm But my line is ok! But sometimes it back sig_analog.c:3807 analog_handle_init_event: Alarm cleared on channel 2 But it again back to red alarm. What can be happen? My lines is all ok! But when I put on Digium Card TDM410 is very inconsistent

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

Would speex be a valid method for compressing/decompressing audio for live communications? Trying to find something that I can use that would decrease the bandwidth requirements tremendously, and still be able to be performed quick enough for live discussion over the air between two communications devices. I have connected two communications devices via the Ethernet, streaming audio via UDP, and

Humm, if I put a filter in this lines, maybe back? 2016-01-05 12:36 GMT-02:00, Ryan, Travis <RyanT at oscarwinski.com>: > >> -----Original Message----- >> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- >> bounces at lists.digium.com] On Behalf Of Vitor Mazuco >> Sent: Tuesday, January 05, 2016 9:21 AM >> To: Asterisk Users Mailing

My analog card, uses a PCI slot and a 12V power connector, is configured with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS ports but I can't dial out from them. Is extensions.conf where I need to make changes? [root at robin asterisk]# cat chan_dahdi.conf [trunkgroups] [channels] [phone](!) usecallerid = yes hidecallerid = no callwaiting = yes usecallingpres = yes