similar to: Problem with SIP-TLS

2015-06-05 12:21 GMT-06:00 Luca Bertoncello <lucabert at lucabert.de>: > Hi list! > > I'm trying to configure my Asterisk to accept SIP-TLS connections, too. > > I followed this HowTo: > > http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/ > > But as soon I try to connect to my Asterisk using SIP-TLS I get on > Asterisk-CLI: > >

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

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

Well, what immediately stands out is: "FILE * open failed!" Have you triple checked that the full filepath is correct and that the user that Asterisk is running as has full permissions to access your valid certificate file? I have it working with microsip and a free TLS cert from LetsEncrypt. When I get to the PC with that on, I can write up what settings I've got if that helps?

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 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

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

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)

Hi, I'm trying to setup an iPhone 4S (iOS 7.1.1) with Linphone to register with TLS to an Asterisk 11.10.0 box. The registration fails and I see this in the Asterisk console: == Problem setting up ssl connection: error:00000000: lib(0):func(0):reason(0) [Jun 8 15:33:39] WARNING[8555]: tcptls.c:274 handle_tcptls_connection: FILE * open failed! Anyone know what that error means? The

On 7/5/19 9:22 PM, Steve Murphy wrote: > hw-- > > I see this kind of behavior when the certificate expires... you've > probably checked this, but sometimes we > miss little details like that. I thought about that and checked the configuration I've been using to create the certificate, and I can't see anywhere that it would expire earlier than after 3650 days. Is

Hi all, i have had sip TLS with an own signed certificate (using the ast_tls_cert script) running on asterisk-1.8.8 - i then have updated to 1.8.9.3 - and now i get the message "FILE * open failed!" I have already recreated the certificates with the script - but still no luck... Does anyone here know the source of the problem ? best regards, Wolfgang Pichler

I'm having some problems with a vega sangoma, if a call comes into my ivr and hangs up, the call continues to ring and leaves hanging the channel, I have to restart Asterisk and everything works Ok my sangoma is a vega 50 , 4 FXO . I tried different tone of countries and does not work, this is the trace of which is for hanging up the channel: http://pastebin.com/y410Rhzt I was thinking

Hi list , this is a bug? ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client regardss -- rickygm http://gnuforever.homelinux.com

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 >

Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com

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)!

2015-03-18 11:13 GMT-06:00 ricky gutierrez <xserverlinux at gmail.com>: > Hi , I'm trying to apply this patch from the source asterisk > asterisk-11.16.0 and when I apply it shows me this message > > asterisk-11.16.0]#patch -p0 < refs > patch: **** Only garbage was found in the patch input. > > is the correct way to apply the patch or am I doing wrong? >

Hi, I know there are people with much experience in asterisk, and I want to ask if anyone had experiance with this gw http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/ I'm having trouble getting connect with asterisk anyone has any production? regardss -- rickygm http://gnuforever.homelinux.com

Zitat von Remko Lodder <remko at freebsd.org>: Hi Remko, > Emails can only be read if they are authenticated / authorized in > someway to access the store. That means you might need to share the > info@ credentials with the other > people so that they can read it over imap or webmail etc. That is self-evident and it is not a problem. I can't understand what you

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)