similar to: Problem with TLS/SRTP with Asterisk 11.8.1

At 09:09 AM 4/18/2016, you wrote: >On Mon, 18 Apr 2016, david wrote: > >>FOLLOWUP & REPORT >> >>I had lots of suggestions, and the most persuasive was to try >>OpenVPN. I already had a CA working, so issuing certificates was >>easy. The HOW-TO guides were less helpful than I could hope, but >>comparing several of them, applying common sense, and

> > >Folks > >I would like to have my windows 7 laptop communicate with my home >server via a VPN, in such a way that it appears to be "inside" my >home network. It should not only let me appear to be at home for >any external query, but also let me access my computers inside my home. > >I already have this working using M$'s PPTP using my home

Hi, After many hours of testing, I've finally tracked down the issue I have been having with dovecot's SSL support. The problem is that the SSL certs result in "TLS handshaking: SSL_accept() syscall failed: Connection reset by peer" errors *if the certificate granted is not granted for client use*. For servers, I normally generate SSL certificates specifically for servers: [

Hi All, I am using "asterisk-11.12.0" version and I am trying to setup secure call (TLS + SRTP) between two extensions and while making a call, I got following error *CLI> == Using SIP RTP CoS mark 5 -- Executing [6004 at from-office:1] Dial("SIP/6003-00000000", "SIP/6004,20") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/6004 SSL certificate

On Mon, 18 Apr 2016, david wrote: > FOLLOWUP & REPORT > > I had lots of suggestions, and the most persuasive was to try > OpenVPN. I already had a CA working, so issuing certificates was > easy. The HOW-TO guides were less helpful than I could hope, but > comparing several of them, applying common sense, and trying things > out, I arrived at a dead-end. Here's

On Tue, 19 Apr 2016, david wrote: > > > > At 09:09 AM 4/18/2016, you wrote: >> On Mon, 18 Apr 2016, david wrote: >> >> > FOLLOWUP & REPORT >> > >> > I had lots of suggestions, and the most persuasive was to try OpenVPN. I >> > already had a CA working, so issuing certificates was easy. The HOW-TO >> > guides were less

This (very quick) patch allows you to connect with the commercial ssh.com windows client and use x509 certs for hostkeys. You have to import your CA cert (ca.crt) in the windows client and certify your hostkey: $ cat << 'EOF' > x509v3.cnf CERTPATHLEN = 1 CERTUSAGE = digitalSignature,keyCertSign CERTIP = 0.0.0.0 [x509v3_CA]

Hello All, I'm having issues joining some Ubuntu servers to an Active Directory domain with trusts. All my machines are running samba and winbind. I have a two domains, we'll call them CORPORATE and CUSTOMER. CUSTOMER has a one way trust with CORPORATE, such that any resources CUSTOMER can access, CORPORATE can as well, but not vice-versa. On all of my CORPORATE machines, users are

Folks I would like to have my windows 7 laptop communicate with my home server via a VPN, in such a way that it appears to be "inside" my home network. It should not only let me appear to be at home for any external query, but also let me access my computers inside my home. I already have this working using M$'s PPTP using my home Centos 6 gateway/router as the PoPToP server.

Hey there i'm trying to get an Asterisk 11.11 with encryption working with my Grandstream phones. But i stuck. To avoid NAT problems i'm using IPv6 Just with TCP/TLS it's working fine. Only the SRTP funktion is not working. Asterisk tells me WARNING[6938]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fa10800f5a0 (len 681) to [2a02:1205::...]:37635 returned -2: Success and also SSL

Today I've been trying to get dovecot (1.0 rc2) to use certificates for client side authentication. If my memory serves right, beta8 had no problems with it (although it was some time ago and on different machine). Similar setup works perfectly well for postfix (for authentication that is, on the same machine). Originally I thought I overdid some certificate settings (keyUsage, nsCertType,

Hi, I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI command to see if SRTP is active on a channel/call. I went through sip show ... and core show channel... and did not see any mentioning of SRTP while there is an SRTP call active. Thanks, Patrick

Hi experts, I am trying Asterisk SRTP in my environment, and find that when Asterisk is behind a NAT, the audi/video UDP ports opened for SRTP relay by Asterisk are local ports on the Asterisk server, media from the two clients out of the NAT (for example from Internet) can not reach the ports, and thus the two client can not establish the secure call via Asterisk. I have set up a STUN server

Hello I will try give you best answer what I can. - alma linux 9, fresh installation, for testing only in virtualbox - packages from Sernet, installad via YUM from oficial repo - installed version 4.18 (same on original linux) - moved backup from original server, /var/lib/samba + /etc/krb, /etc/default/samba, /etc/samba - original domain created if I remember on 4.15 or 4.16, then schema

I have used every asterisk 11.8.X version. Have not had an issue till 11.8.0 and 11.8.1 When I use ConfBridge I am attempting to put all participants in MUTE mode and just one talker... However, since 11.8.0 I am hearing feedback in the announcement like the channel is not really muted. I dropped back to 11.7.0 and I hear no feedback. Has something changed - or - am I not correctly setting up

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1, and 12.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolve

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1, and 12.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolve

Hi, I?m trying to get TLS to work with asterisk and client phones, and all I?m getting from asterisk is [Aug 23 11:46:42] WARNING[1170]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) [Aug 23 11:46:44] WARNING[1171]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! when clients try to

Thanks for getting back to me. Sadly I've not had the time today to attempt the reproduction. Can you, just to save me time, double-check if this happens on a server with the Samba 4.20 being a just from-our-tarball Samba and show the logs that gives? Thanks, Andrew Bartlett On Wed, 2024-04-10 at 10:04 +0000, Tom?? Havl?n via samba wrote: > HelloI will try give you best answer what I can.

We're experiencing an issue where calls disconnect after 15 minutes. It seems to happen just after Asterisk sends an update mesage. RTP is being set up directly. Asterisk is only in the SIP dialog. Has anyone experienced this issue? 4 PRIs inbound, 4 PRIs outbound, asterisk provides switching. SIP/2.0 200 OK Via: SIP/2.0/UDP 38.XXX.XXX.XXX:5060;branch=z9hG4bK1c4b524f From: