similar to: [somewhat OT] seeking ideas/input for my thesis

Hello everyone, I am developing an online SMDR / call log system for asterisk. This is going to take the form of an executable with embedded sql and webserver, pdf generation, excel generation, graphs. Actually, we have been selling this for a while now with great success and now I am starting work on the integration with Asterisk. Its a windows executbale and the executable is just about 1MB.

Hi All, I've been reading about Phil Zimmermann's ZRTP encryption scheme for SIP clients. This seems attactive but I don't use soft phones. I'm guessing that we'd need ZRTP support in Asterisk in order to use it to secure calls from hard phones. There seem to be issues with SRTP key exhange between various devices. So much so that the IETF is working on a standardization

Has anyone implemented HOBIC SMDR output from *? Can someone point me to the Bell HOBIC specification? Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Hello, I am searching for a solution to encrypt authentication from Asterisk server to clients. Searching srtp seem to encrypt traffic, I just want client authentication with encryption. Can someone point to the right direction? has anybody used ZRTP? experience with ZRTP? Thanks, _motty

Dear All, I seem to have stumbled across an AGI problem; I have written an AGI Script (bottom of this email); The script does the following; Makes a CDR entry when called Records the call Updates the CDR Finds a corresponding DNIS from the SMDR table (captured via a serial port logger) Matches up the record and updates the CDR. The script works perfectly in my test lab and has been doing so

On 10/28/2015 06:37 PM, Pete Mundy wrote: > Hi Motty, > > Isn't the whole point of the nonce in a SIP registration to ensure the > secret doesn't go on the wire in plain-text? Is this not enough, or > are you looking to hide the username too? > > (if so, fair 'nuf, just wondering why :) > > Pete > > Ps, if so then I think TLS is the missing part of

Hello. Do I understand correctly that the current implementation res_pjsip does not support ZRTP? http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html Nothing has changed since 2013? P.S. I greatly regret that moved from chan_sip to res_pjsip. Previously used very much lacking, and much of the promise failed. Dmitriy Serov. -------------- next part -------------- An HTML

Hello List I am very interested in developing a research project on security protocol for VoIP, under the GPL. For some time I have been reviewing ZRTP, I would like to know the opinion having regard to whether and under asterisk, but I see that this closed implementations according am Http://bugs.digium.com/view.php?id=10024 Are Zphone and ZRTP the future for the Voip Security? Opinions?

Folks! Let me clarify something to the Asterisk community about the CDR tool. 1) This is *not* my code to start with. I picked the original code from this forum here... http://www.voip-info.org/tiki-index.php?page=Asterisk%20CDR%20Areski%20GUI 2) The original code was not working (for most part, as the MySql portion has bugs) and I fixed this and added a few bells and whistles. 3) The

05.10.2015 23:24, Joshua Colp ?????: > On 15-10-05 05:22 PM, Dmitriy Serov wrote: >> Hello. Do I understand correctly that the current implementation >> res_pjsip does not support ZRTP? >> http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html > > ZRTP is not supported in Asterisk itself. > >> Nothing has changed since 2013? P.S. I greatly

What's the status of ZRTP supported by Asterisk? There was some discussion on the -dev list and -users list, but it was inconclusive. At about the same timeframe, a bug (#0010024) was opened and updated for several months, but has been "suspended" since late 2007. Does any version (1.4.x, 1.6.x) of Asterisk support ZRTP with clients (or with other servers)? Any successful testing

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:

Ooops did Google just become a carrier :) http://googleblog.blogspot.com/2007/07/all-aboard.html I hear stocks crumbling worldwide as I type. Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070703/92218fc6/attachment.htm

Dear all, I have Asterisk 1.4.13 and I need to use encryption among Asterisk and my SIP users, and with the RTP data interchanged among users. I prefer the use of ZRTP/SRTP because we use Twinkle and X-Lite/Zfone as our voip clients and they support these encryption mechanism. My question is: do I have to enable any encryption support in Asterisk 1.4.13 ??? Or Asterisk has native encryption

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, A potential client (hotel) has a Property Management System that talks the "Mitel" protocol to their current Mitel PBX in order to receive CDRs (which end up being rated by the PMS system and charged back to guests). Does anyone know of any (free or otherwise) docs on this protocol, or better still have experience interfacing asterisk in a hotel situation like this? The PMS

Friends, The following mail was sent earlier to asterisk-dev and did not cause the amount of discussion I hoped it would. Now that we have a way to secure signalling in IAX2 and SIP in Asterisk svn trunk, we need to start working on the concept of a "secure call" - or does it really matter? In SIP, there's a specification for how I as a domain owner can request all calls to

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I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

Dear people, does anybody try the ZRTP patch for Asterisk in order to have ZRTP encrytion among SIP/RTP calls ??? In other words, did anybody succesfully implement ZRTP in Asterisk ??? Any documentation about it ??? Special thanks Alejandro