similar to: SRTP only from asterisk to extention possible

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

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

Am 03.03.2015 um 18:16 schrieb James B. Byrne: > 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

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

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

On 26-10-16 15:03, Dan Jenkins wrote: > > > On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > > I keep getting the following error when trying to connect to the > Asterisk server using AMI : > > $socket = fsockopen("tls://11.22.33.44 >

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Everybody, I am experiencing some troubles with my Bria iPhone Edition (v. 1.2.8 build 5312, on iOS 4.2.1 iPhone 3G) and Asterisk 1.8.3.2 + TLS/SRTP on LAN (without NAT). With 2 computer clients (Blink, one on Mac, one on Windows/Linux),9i can have a very fine secure conversation in both directions. When I want to do the same with my iPhone,

Hi, I followed the TLS/SRTP tutorial on the wiki [0] using Asterisk 11.8.1 on CentOS 6.5 x86_64 and CSipSimple on a Nexus with Android 4.4.x local wifi. The phone seems to register but directly after that things fall apart (turning SELinux off made no difference): *CLI> -- Registered SIP 'encrypted' at 10.0.0.137:58079 > Saved useragent

I'm having trouble setting up tls/srtp secure communications on my Asterisk server- I'm still rather new at working with Asterisk. I have enabled tls and encryption and I have csipsimple with tls build on the phone. I'm currently only testing one phone with this capability so far, and the rest still work in the current state. My logging looks like this with verbose turned up:

Hi all, We're testing TLS and SRTP on Asterisk 1.8.10.0 and have it working with a commerical (not self-sign) AlphaSSL wildcard (GlobalSign) using Blink Lite 1.6.2 as per https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial We've tested with Bria on an iPhone and that doesn't recognised the commercial CA (GlobalSign Root CA). On a Yealink 28P with V60/V61 is registers

On 7/6/19 10:40 AM, Michael Maier wrote: > On 05.07.19 at 22:02 hw wrote: >> >> openssl verify -CAfile ca.pem asterisk.pem >> asterisk.pem: OK >> >> >> When I set tlsdontverifyserver=yes, it works (i. e. asterisk registers >> to the SIP provider and there is no error message).  Otherwise I'm >> getting the error message and asterisk does not

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 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]

Hi, when trying to use SRTP, I can see UDP traffic from phones to the asterisk server being dropped be the firewall on arbitrary ports. Where do I configure the SRTP port range (like the rtp port range)? Why aren't the clients talking to each other directly but apparenty try to send the SRTP traffic to the server? That the traffic is being blocked by the firewall is probably the reason

I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.org I followed this guide: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages The SRTP support appears to be missing though. I notice libsrtp was not automatically installed as a dependency, and no srtp module exists under /usr/lib64/asterisk/modules Is it still necessary to do a source build every time SRTP is

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

please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle, ...) --------------------------------------- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA - http://lcna.slu.cz =======================================