similar to: Asterisk 1.8 TLS with Softphone blink on Windows don´t work

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

Microsip (Windows) is free and small. 2.5Mb download, 10Mb RAM usage, does everything I need and configuring TLS is a doddle. http://www.microsip.org/ On 16 February 2017 at 13:04, Max Grobecker <max.grobecker at ml.grobecker.info> wrote: > Hello, > > I'm a big fan of PhonerLite. > It's more poplar in Germany, but also available in English language. > This client

Hi, I tried it the implementation of TLS in asterisk 1.8.4.3 on ubuntu 10.04. I follow the tutorial: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial. and I use blink as a softphone in ny client in windows. for regular communication process (without TLS) smoothly, but when it just follow the tutorial, it is always error on his softphone: transport error. my configuration like

Does anyone know of a reputable repository where I can obtain a CentOS-6 version of Blink as an rpm/yum package? I have to support three platforms (OSX, MS-Win and CentOS-Linux) with SRTP capable soft-phones. interoperating with an Asterisk-11 PBX; and I would rather have only one client to deal with. -- *** E-Mail is NOT a SECURE channel *** James B. Byrne

Author: prasad Repository: /hg/zfs-crypto/gate Revision: 9b81f2d852bc79f96bcc4e7ac82d260561b25984 Log message: 6374127 Solaris PCIE hotplug should not blink green LED during ''configure'' operation Files: update: usr/src/uts/common/io/hotplug/pciehpc/pciehpc.c

Hi, I'm excited to announce that the guys from AG Projects are stopping by for a beer tomorrow on VoIP Users Conference, aka VUC. You should already be familiar with their excellent multi-platform SIP client, Blink (http://icanblink.com) While Adrian and Sa?l enjoy a few exotic brews with us, they'll also be telling us about SylkServer: - creation and delivery of rich multimedia

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

Hello, I am using asterisk 14.2 and PJSIP, with TLS transport. I?m sure I?m doing something wrong here .. In 2 distinct softphone clients (Bria and Groundwire), I am able to register successfully, and place a SIP call, with no certificate warnings. But shortly after I place that first call and hang up, I receive a certificate name mismatch error in the softphone, the error presenting me

Hello; I have asterisk?Asterisk 1.8.23.0-vici and Polycom 331 and I am able to register from local area network and not able to register from outside the office. Also from outside the office, I am able to register via PhonerLite softphone and not able to register via Zoiper softphone. So from outside the office, I am not able to register from Zoiper softphone and not able to register from

Hi, I have written an click2dial application that rings an agent soft phone and connects the agent with a customer. very often I can see, that the agent softphones signal a busy back to server, although the phone is definitely hung up and the previous calls where handled normally. I testet 3cx Version 6 and X-Lite V1 up to V3... all show the same misbehaviour. I did a SIP Trace and can see,

Hi Karsten, On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote: > .... The server sends Server Hello, Certificate, Server Key > Exchange and Server Hello Done. Something in that packet seems to be unacceptable for openssl 1.1.1d as it is compiled and configured for Buster. Certificate length, Digest algorithm, ... You my change the system default settings at the

Hi, I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client hello from Asterisk to the server. The server sends Server Hello, Certificate, Server Key Exchange and Server Hello Done. Than Asterisk sends back a Alert (Level: Fatal, Description Handshake Failure). The following line appears in the log: ast_iostream_start_tls: Problem

I''m trying to use Rails on Windows 2000. I''m running into an immediate problem in that all the files generated by Rails use the UNIX end-of-line convention instead of the Windows end-of-line convention. A non-compliant end-of-line convention will break any number of tools I may wish to use to manage these files. Is there a way to get Rails to emit Windows-compliant

I've noticed in the last day that I'm not getting any Bugzilla email any more. I haven't changed my settings in Bugzilla. Did something change at UIUC? Reid. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20031118/ca01ce8a/attachment.html> -------------- next part -------------- A non-text

On Mon, 17 Nov 2003, Reid Spencer wrote: > I've noticed in the last day that I'm not getting any Bugzilla email any > more. I haven't changed my settings in Bugzilla. Did something change > at UIUC? I don't think so... I'm still getting plenty of bugzilla mail. :) -Chris -- http://llvm.cs.uiuc.edu/ http://www.nondot.org/~sabre/Projects/

I have been sending OPTIONS requests 1) programatically (my own code), 2)manually via SIP VERIFY PEER x and 3)automatcially by setting verify=yes in sip.conf. The trouble is I do not see anything except an ACK 200 come back from endpoints and it does not contain any SDP/codec info. . My goal is to determine audio and video codec capability in advance of a call INVITE. I notice in both 2 and 3

Hello, Could someone with the requisite shorewall expertise please help me? Here is a description of my problem. I dial in to my ISP using kppp. It seems to establish a connection just fine. However, only a handful of bytes are exchanged. I must then become ''root'' and issue ''shorewall start'' in order to get the Internet connection to work normally. Once

Hello, I am reasoning about a question concerning the t-test for one sample. My data includes 150 values (mean 10.07) which I want to compare to mu=9. A tow-sided t-test yields > t.test(data,mu=9) One Sample t-test data: data t = 3.0099, df = 149, p-value = 0.00307 alternative hypothesis: true mean is not equal to 9 95 percent confidence interval: 9.368676 10.777991 sample estimates:

HelloI have the following system Asterisk 1.6.1dahdi 2.2.0.2 TE420P card Centos I have noticed that all the four lights are blinking(ie coming red and then off so on)... Previously I also noted that when dahdi drivers are not installed lights blink but one by one in sequence(like in marriage cermonies :P) and after dahdi installation lights get off ... but this time all at same time

Hello! Just got new server with TE420. Not all four spans will be used immediately, but spans not configured or not connected blink red light. Is it possible to turn span off, so my colleagues will not eventally tell me that something is wrong with asterisk? :-) Thank you!