similar to: Improve Wiki's "WebRTC config" page

In the asterisk wiki instructions for Configuring Asterisk for WebRTC clients... https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients "To communicate with websocket clients, Asterisk uses its built-in HTTP daemon. Configure /etc/asterisk/http.conf as follows: [general] enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089

hi, i'm fighting with webrtc for 14 days reporting my experience to minimize number of crazy asterisk users i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip + secure websockets + secure audio + audio in both ways problems first, i needed run chan_sip for old hard phones and wss with chan_pjsip only for webrtc. this is possible with patch from

Hi Alexey, This is what works for me: [http.conf]: tlsenable=yes ; enable tls - default no. tlsbindaddr=144.x.y.z:8089 ; address and port to bind to - default is bindaddr and port 8089. tlscertfile=/etc/asterisk/keys/mycert.pem ; path to the certificate file (*.pem) only. tlsprivatekey=/etc/asterisk/keys/mycert.pem ; path to private key file (*.pem) only. Date: Tue, 13 Jan

Hello, Le lun. 6 janv. 2020 à 19:01, Olivier <oza.4h07 at gmail.com> a écrit : > May I add I could successfully (if pjsip show transports has any meaning) > add a PJSIP TLS-transport with: > > [transport-tls] > type=transport > protocol=tls > bind=0.0.0.0:5061 > cert_file=/etc/asterisk/keys/asterisk.crt > priv_key_file=/etc/asterisk/keys/asterisk.key >

Hi, I'm trying to connect to the asterisk pbx via wss, from sipml5.org demo page (http://sipml5.org/call.htm). I used the guide from https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial , to setup the tls. I could make a secure sip call ( SRTP) using the PhonerLite sip client. ( This confirms my sip - tls settings and tls certficates. ( I'd added the tls client certficate

Hello, On a newly re-installed Asterisk 16.7.0 on Debian Buster, I can't find a way to enable HTTPS. Asterisk is running as asterisk:asterisk: asterisk 11097 0.3 6.7 741352 67984 ? Ssl 17:53 0:06 /usr/sbin/asterisk -g -f -p -U asterisk # cat /etc/asterisk/http.conf [general] servername=Asterisk enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089

CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten

Hello, After countless hours on, this I found the root cause of HTTPS settings on Debian Buster. All this came from ast_tls_cert script using 1024 bits-long keys where Debian's defaut was to require at least 2048-long keys ! Simply passing -b 2048 to ast_tls_cert solved it. 1. May I suggest mentioning explicitly this possibility in wiki page [1] ? 2. What would you say of adding an extra

On Tue, March 3, 2015 13:37, James Cloos wrote: >>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: > > JBB> tcpenable=yes > JBB> tlsenable=yes > JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt > JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt > JBB> tlsdontverifyserver=yes > JBB>

>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: JBB> tcpenable=yes JBB> tlsenable=yes JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt JBB> tlsdontverifyserver=yes JBB> tlscipher=ALL JBB> tlsclientmethod=tlsv1 You are missing the tls key. The config name is

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'm having the same issue! The difference in my case is Asterisk server has a public IPv4 and the browser is behind a single NAT. I'm forwarding my configuration below (which I posted previously on asterisk-users). How can we debug ICE negotiation? ---------- Forwarded message ---------- From: Vinicius Fontes <vinicius at aittelecom.com.br> Date: 2015-07-27 13:54 GMT-03:00

May I add I could successfully (if pjsip show transports has any meaning) add a PJSIP TLS-transport with: [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cert_file=/etc/asterisk/keys/asterisk.crt priv_key_file=/etc/asterisk/keys/asterisk.key method=tlsv1 Le lun. 6 janv. 2020 à 18:33, Olivier <oza.4h07 at gmail.com> a écrit : > Hello, > > On a newly re-installed

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

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>

Hello, I am trying to secure SIP session with TLS on Asterisk Server 1.8. I keep getter an error, == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection: FILE * open failed! I tried both signed and self-signed cert to no avail. Here is my Configuration: Sip.conf

Hi, I have to connect 3 asterisk servers,each of them being TLS server for his clients and connected in both way in TLS with both others asterisk, each having hi own Common Name. Is this possible? I set up 2 asterik's , one server and the other client, this is OK. But I can't deal with certificats generated on both servers. I tried to put tlscertfile ans tlscafile in the peer

Hello I keep getting the following error when trying to connect to the Asterisk server using AMI : $socket = fsockopen("tls://11.22.33.44","5039", $errno, $errstr, 5); Erorr on CLI : [Oct 26 14:38:19] ERROR[2992]: tcptls.c:609 handle_tcptls_connection: Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [Oct 26 14:38:19]

2016-02-19 12:01 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > on my own server > Today, I'm back from holidays trip. First of all, thanks for replying ! I'll try to use jssip as you suggested. Anyway, I'm still failing to understand if wiki's page [1] is still valid with Asterisk 13, and if it's not valid anymore, which is the main change that prevent

Hello, I've installed a new Asterisk 17.0.0 on a Debian Buster system. This Asterisk instance is run by asterisk user (and group). I've got: # ls -l /etc/asterisk total 68 -rw-r--r-- 1 asterisk asterisk 501 nov. 18 19:12 asterisk.conf -rw-r--r-- 1 asterisk asterisk 135 nov. 18 18:57 cdr.conf -rw-r--r-- 1 asterisk asterisk 684 nov. 18 18:57 cdr_custom.conf -rw-r--r-- 1 asterisk