similar to: Restricting concurrent SIP calls

Hello, today I found this security report regarding Asterisk SIP Security. http://www.securiteam.com/securitynews/5LP0720B5G.html Maybe It could help somebody who isn't using a newer than 15th of August cvs version. Best regards Lubo

Hello, There is a new asterisk vulnerability report at this address: http://www.securiteam.com/unixfocus/5HP0H1PB5S.html This is the second security report regarding asterisk for 8 days (http://www.securiteam.com/securitynews/5LP0720B5G.html) Both fixes was reported and fixed silently. My question is: Is it possible in the future such a security problems to be reported in this mailing list

Hi Everybody, I am new to Asterisk. Can anybody suggest me some link where I can find architecture level detail of this system. My aim is to find out how easy it is to port it on a new hardware (T1/E1 and POTS)? Any input is highly appreciated. Regards Arun

Here's another interesting problem: if you recall I have a data frame (LCvars1) that consists of about 1500 cases (rows) of data from kids who took a test of listening comprehension. The columns are their scores (1 = correct, 0 = incorrect, . = missing) on 140 test items. The items are numbered sequentially and are ordered by increasing difficulty as you go from left to right across the

Hello, after upgrading my asterisk from cvs few days ago I discovered that all my anddefault asterisk voice messages/prompts was played wrong - they was played so slowly ... and the IVR system wasn't usable at all. Today I upgraded everything: a fresh cvs copy, made make install and make samples - but after that the problem with IVR was the same: ALL default asterisk sounds in gsm format

Hello every one, I have got a H323 gatekeeper for testing. The informations are something like this: account code: test01 gk ip address:192.168.10.12 I don't know how to set it in the h323.conf or oh323.conf, I have tried it for almost one day but I always got the error. Help me please. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Today class we are going to be talking about the wonderful line of GrandStream products. Or should I say BarbieTone phones? Who else is having MAJOR issues with the grandstream products? How many times have you been told upgrade upgrade upgrade? How many of you have paperweights, granted the phone is light as a feather and couldn't weight papers down in the first place? How about that

Hi everyone, We just upgraded Samba from 4.4.5 to 4.6.5 and appear to be experiencing a problem with authentication, when the RPC domain is not supplied as part of the username. I have two scenarios where this has cropped up: RADIUS authentication using ntlm_auth Apache HTTP using mod_auth_ntlm_winbind RADIUS authentication: We use the freeRADIUS 'mschap' module to provide

Anyone have any thoughts on why versions of asterisk I try (4 so far) after CVS-07/18/03 always end up locking up on me... which means no sip clients can register/re-register and if I type "reload" or "stop now" at the cli it just returns and does nothing. I have experienced this same issue on three separate boxes. Two running RedHat 9 and one running Redhat 8. I don't

I have apparently found an error in the "pam" function of the "cluster" library package. Please pardon me if this error has been pointed out or if this e-mail should be directed to someone else. The problem only started occurring with R version 1.5.0, which I started using about a week ago. The problem occurs when you try to use "pam" with the input being a

Here's the problem I'm trying to solve in R: I have a data frame that consists of about 1500 cases (rows) of data from kids who took a test of listening comprehension. The columns are their scores (1 = correct, 0 = incorrect, . = missing) on 140 test items. The items are numbered sequentially and are ordered by increasing difficulty as you go from left to right across the columns. I want

Is it possible to limit the number of "lines" provided by a given SIP/IAX connection? For example: I want to limit SIP extensions to only a single incoming line, even the phone itself can handle three. Or, I might want to prevent extensions from making more than one outgoing call at a time. Or, I might want to protect my bandwidth/call quality by limiting outgoing calls through

Hi folks, As none of the SIP softphones that I tested can disable more than one incoming call, I decided to implement it by software ;-) I'm attaching a patch that does it. To make it work, modify your sip.conf file and include callwaiting=[0|1] at the general section, or for each peer that you wish to control. Please note that I haven't tested it too much, and my source tree is quite

HI ALL; Is there anybody who use app_radius(astersik radius module)??????????? is it stable? Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040803/8a096bfe/attachment.htm

Hi I''m pretty much a newbie to Rails and I have at least two projects under development that are going to need the same type of constructs and I''m struggling to get my head around how to achieve what I want using Rails models and relationships. In one project I have a hierarchy of (geographical) sites. For example: Earth Europe Ireland Cork Kerry Galway

Merry Christmas from the Colorado Organization for Victims' Assistance. Our (Comdial) PBX fried after a power failure. Thanks to Mark Spencer, Digium, VCCH, and the friends who support this group, we are now back "on the air". We wish everyone good health for the coming year.

Hi! All I would appreciate if someone could advice me on how stable is sip-h323 & h323-sip translation as well as how many calls can it handle when doing such translation.( assuming single 2.8Ghz intel processor & 1GB RAM) Regards, John -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

Hello, I just want to ask if it has been considered to use OpenSSL for the encryption stuff. I guess they've spent quite some time on the subject and have the cipher routines both debugged and optimized. I know that some of the current cipher code was taken from SSLeay but since OpenSSL is de facto SSLeay successor, it might give tinc more flexibility and allow the user to choose

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

I'm looking for a prepaid calling card platform that: * easily scales to multiple servers with a common database for: redundancy, capacity, and performance Looking to start with capacity to handle 100 simultaneous calls and be able to easily scale to 1000+ simultaneous calls. * in addition to the normal anti-fraud measures, supports an API for easily adding new anti-fraud tests along the