similar to: Cannot configure PJSIP TLS

Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP. tcpdump on asterisk server showing UDP packet bound for my remote endpoints internal IP: 17:07:57.130212 IP

Hi, after switching from chan_sip to chan_pjsip, a device running Grandstream Wave leads to the following error message on the asterisk console: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> <SSL routines- ssl3_get_client_hello-no shared cipher> len: 0 peer: 10.10.20.29:43357 Something with the encryption must have changed with asterisk. How can I get the device to

On Thursday, January 23, 2020 11:31:46 PM CET Sean Bright wrote: > On 1/21/2020 9:18 PM, hw wrote: > > [transport-tls] > > type = transport > > protocol = tls > > bind = 0.0.0.0:5061 > > tos = cs5 > > cert_file = /etc/asterisk/cert/asterisk.pem > > ca_list_file = /etc/pki/tls/certs/ca-bundle.crt > > method = sslv23 > > This is what mine

Thanks, Michael. A few questions: Is [transport_name] a reserved word, or am I supposed to replace it with a name of my own, like '[did-transport]'? Some of the keywords I haven't seen before. Is ca_list_file supposed to be an aggregate of the public and private key? And what are the 'method,' 'tos' and 'cos' keywords, which are commented out in your

Hello Steve, use the following configuration for the transport and bind this transport to the trunk: [transport_name] type=transport protocol=tls bind=192.168.13.24 ; your bind IP ca_list_file=/etc/pki/tls/certs/ca-bundle.crt ; method=tlsv1_2 verify_server=yes allow_reload=no ;tos=0xb8 ;cos=3 external_media_address=your.ext.host.name ; hostname pointing to your ext. IP

Stuck with TLS transport, I have 2 phones (both in local network for tests) one connected with up second with tls when I calling TLS to UPD everything is fine, but when UDP calls TLS I getting an error ERROR[44230]: pjsip:0 <?>: tlsc0x7f143012 TLS connect() error: Connection refused [code=120111] pjsip log: -- Called PJSIP/601/sip:601 at 192.168.1.55:5075;transport=tls <---

On Saturday, April 18, 2020 5:42:11 PM CEST Joshua C. Colp wrote: > On Sat, Apr 18, 2020 at 8:47 AM hw <hw at gc-24.de> wrote: > > Hi, > > > > how do I make a bug report? I filled in the form to make a report and > > https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues > > reported by me. > > If successful then JIRA will redirect

Greetings ! My goal is to get Twilio trunking working, and with TLS/SRTP. I see this concerning message in my log: [Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object of type 'endpoint' with id ?twilio' from configuration file ?pjsip.conf? Thus, ?pjsip show endpoints? does not show the endpoint for the Twilio trunk. Hoping for a sanity check of

I want to configure communication with my phone provider using TLS for all the obvious reasons. Since I'm behind a firewall, I'll be needing to do it with NAT. There are examples of UDP plus NAT in pjsip.conf, but none for TLS plus NAT. Would it be correct to set up the TLS transport stanza to look like the [transport-udp-nat] stanza example, replacing UDP with TLS in lines like

Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call

Hi, how do I make a bug report? I filled in the form to make a report and https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues reported by me. If someone knows how to get asterisk to re-register when using pjsip after the registration shows as Rejected, like after the internet connection to the VOIP provider goes away (and comes back), please let me know. This bug makes

On 1/21/2020 9:18 PM, hw wrote: > [transport-tls] > type = transport > protocol = tls > bind = 0.0.0.0:5061 > tos = cs5 > cert_file = /etc/asterisk/cert/asterisk.pem > ca_list_file = /etc/pki/tls/certs/ca-bundle.crt > method = sslv23 This is what mine looks like which works just fine: [transport-tls] type          = transport protocol      = tls method        = tlsv1_2

Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed to make it work, all my terminals spa Cisco 5XX look my cli [Jul 8 11:09:16] ERROR[14733]: pjsip:0 <?>: tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:16] WARNING[14733]: pjsip:0 <?>: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)!

Hi first post, so hope I'm not violating protocol. Been using Asterisk as home phone and hobby use for nearly 10 years. I love this project. Anyway, would someone mind verifying my pjsip.conf ? This seems to work well for 14.3.1 but I get no rtp into my natted Linphone when I upgrade to 14.4.1. Other than that the phone registers properly on 14.4.1. I can provide a pjsip log as well,

Hi guys, have really annoying problem with trunks when I calling over voip provider.. after awhile provider sends INFO packages but for some reason Asterisk doesn?t answer on it. after 8 packagers provider just drops the call, here is the package <--- Received SIP request (555 bytes) from UDP:192.168.53.1:5060 ---> INFO sip:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9 at 192.168.53.9:5060 SIP/2.0

On 1/23/2020 6:04 PM, hw wrote: >> This is what mine looks like which works just fine: >> >> [transport-tls] >> type = transport >> protocol = tls >> method = tlsv1_2 >> cipher = >> ECDHE-ECDSA-AES256-GCM-SHA384,ECDHE-RSA-AES256-GCM-SHA384,ECDHE-ECDSA-AES128 >>

I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is behind a NAT router, the physical setup is very much a trivial one. The Asterisk PBX is supposed to act as the telephone gateway for several VoIP/SIP phones. I'm using throughout pjsip as configuration, I have no experience with chan_sip since I started recently using Asterisk for several SoHo and lab's

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

Hello All, I have an Asterisk server v13.1.0 running on EC2 and I am able to connect and register SIP devices and "see" them on the asterisk CLI. I am also able to place calls, but I am not able to hear any audio on either end after the call is picked up. I was wondering if you can tell me what a minimal configuration for Asterisk on EC2 looks like. My current pjsip.conf configuration

Scenario: Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to 192.168.254.1:5060) is behind a NAT, acting as a client to our ITSPs SIP server. But also, this Asterisk is server for various VoIP telephones. Acoording to Asterisk's wiki, the transport section of pjsip.conf is configured as follows: ; Transport via UDP [transport-nat-udp] type= transport