similar to: Client TLS certificates for auth ?

What is the best virtual server tech (and most stable, etc) to use for a asterisk virtual hosting environment? I have a client that wants to do virtual hosting of Asterisk (only SIP or IAX, no PRI, etc) and I'm wondering if Xen or something else would be best? We'd like to stay away from the costs of VMWare if possible. Thanks! Travis -------------- next part -------------- An HTML

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

Hi folks, At least several endpoints (soft phone and desk phones) are supporting various 256 bit ciphers for SRTP these days. I *believe* libsrtp has been updated to allow this, and that only the code in Asterisk has not been been updated to allow these stronger ciphers. Would anyone with the know-how be willing/able to submit a patch ? Thank you, Kevin Long

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

Hi. I have an Asterisk Server (A) where it acts as the main gateway to offer services. There are different asterisk servers (B -D) that connect as extensions to the Server A. I would like to implement TLS and SRTP for these extensions, but have the non secure as well for other extensions. for example the extensions 4500-4504 be with TLS/SRTP and the rest be non secure(ordinary). Is there a guide

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>

One of my clients today asked me about TLS support for encryption of SIP payloads, and I didn't have an adequate answer as to why it wasn't supported or even discussed. Some archive searching finds scant mention of this in reference to Asterisk. Of course, encrypting the SIP payload is only 1/2 the problem; the payload itself is the next problem. I understand that IAX solves these

ricky gutierrez <xserverlinux at gmail.com> schrieb: > Hi lucas , dou you try this: > > https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Tested right now. Same problem... I think it is a problem on Asterisk for OpenWRT... :( Regards Luca Bertoncello (lucabert at lucabert.de)

Hey guys,tried to make tls work with pjsip on asterisk 13.2.0 have compiled pjsip with ssl, added transport [tls] type=transport cert_file=/pbx/keys/server.crt ca_list_file=/pbx/keys/ca.key priv_key_file=/pbx/keys/server.key protocol=tls bind=192.168.1.4:5061 local_net=192.168.1.0/24 external_media_address=77.77.77.77 external_signaling_address=77.77.77.77 have configured Grandstream GXP1400

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

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 I'm just about to upgrade to version 1.8.29.0 and have compiled with SRTP. However, we exclusively use the asterisk realtime architecture using the mysql connector. Looking at tutorials we have to set encryption=yes and transport=tls for any peer we want encrypted traffic for. Having a look at contrib/realtime/mysql/sippeers.sql from the source code shows that the encryption column is

Hello all, I want to use Asterisk with the SRTP patch from http://bugs.digium.com/view.php?id=5413 . I'm confused to create the certificates for it. Can anybody help in such question? P. S. I've created the pem files and renamed it to * ${astetcdir}/asterisk.crt * ${astetcdir}/asterisk.key * ${astetcdir}/ca-certificates.crt but the asterisk got "segmentation fault" error at

Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit

On Sunday, January 26, 2020 3:58:31 PM CET Pete Biggs wrote: > > what does Centos 7 do with UPD packets having invalid checksums? > > By default I assume they are just dropped - that's what should happen. Hm that's what thought. > > Are such packets inevitably dropped? > > Applications can specifically disable checksum checking for the kernel > network stack

Hello, I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLS authentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authentication enabled? Any suggestions? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.long at haloprivacy.com> wrote: > > Thanks George I appreciate the info . Being able to see what codec is in > use for call in progress is very handy sometimes. > > As far as the RTP stats goes, I see there is some info with ?rtp? and > ?rtcp? commands which can be useful for troubleshooting. A running tally of > #

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

Hi, I use Asterisk 11.5.1 and it works fine. :) Now I want to use TLS and media encryption. I followed this guide: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial When I place a call via Blink-Client (0.5.0) I get connected and Blink shows 2 locks. The blue lock shows "Signaling is encrypted using TLS" and the orange lock shows "Media is encrypted using

Hi list, I'm trying to create an asterisk 1.8 rpm with SRTP. I found mention of a libsrtp rpm, <http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm > in these instructions, <http://www.voip-info.org/wiki/view/Asterisk+SRTP> but it is unreachable (by me, anyway). The libSRTP source is here, <http://srtp.sourceforge.net/download.html>. Has this already been packaged for