Asterisk Project Security Advisory – AST-2012-006 Product Asterisk Summary Remote Crash Vulnerability in SIP Channel Driver Nature of Advisory Remote Crash Susceptibility Remote Authenticated Sessions SeverityModerate Exploits Known No Reported On Ap..
In the Skinny channel driver, KEYPAD_BUTTON_MESSAGE events are queued for processing in a buffer allocated on the heap, where each DTMF value that is received is placed on the end of the buffer. Since the length of the buffer is never checked,..
A user of the Asterisk Manager Interface can bypass a security check and execute shell commands when they lack permission to do so. Under normal conditions, a user should only be able to run shell commands if that user has System class authorizati..
Anam It seem should be work, but I just have a question about chan_dahdi.confregardless to parameter rxwink=300; Atlas seems to use long (250ms) winks By the way gain parameters shouldn´t have any effect to CID signal processing, how about to comme..
Group, is in MeetMe any option to identify the own number (from the view of a caller)? I would like to write an option to set on the telephone an request for voice, if the room is muted. That request should display on our Conference Control Website ..
This is a very strange problem (at least for me). I just realized that started from April 20th 2012 every inbound call is from unknown. Prior that, asterisk succesfully displayed the caller callers ID for SOME of the calls (30-50% success rate). I..
In DAHDI 2.6.1 changelog includes this : README, drivers/dahdi/Kbuild: Build OSLEC EC if in the tree Build the OSLEC echo canceller (drivers/staging/echo and dahdi_echocan_oslec) if the code of oslec is present in the tree. Also closing another is..
, I am looking for an open source speech recognition engine for a hobby project. There used to be a Sphinx interface for the generic speech API (http://scribblej.com/svn/) but it does not compile on Asterisk versions later than 1.6.x Could anybody dir..
Hi I have a small problems with incoming call. I have a peer actually configured for outcall : sip.conf: [Trunk-Telco] type=peer host=domaineofmysupplier.net outboundproxy=domaineofmysupplier.net session-timers=originate session-expires=7200 qualify=..
To put it simply, is the process where Asterisk tries to redirect the RTP media stream to go directly from the caller to the callee. Be careful that some devices do not support this (especially if one of them is behind a NAT). The default setting..