similar to: AST-2011-001: Stack buffer overflow in SIP channel driver

Asterisk Project Security Advisory - AST-2011-006 Product Asterisk Summary Asterisk Manager User Shell Access Nature of Advisory Permission Escalation Susceptibility Remote Authenticated Sessions Severity Minor

Greetings, Recently a performance regression in chan_sip was discovered in Asterisk 1.8. The regression is caused by chan_sip setting MASTER_CHANNEL(HASH(SIP_CAUSE,<chan name>)) after each response received on a channel. That feature has been made optional in the latest 1.8 SVN code, but is currently still enabled by default. After some internal discussion, we decided to consider disabling

The Asterisk Development Team has announced security releases for the following versions of Asterisk: * 1.4.38.1 * 1.4.39.1 * 1.6.1.21 * 1.6.2.15.1 * 1.6.2.16.1 * 1.8.1.2 * 1.8.2.1 These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, 1.8.1.2, and 1.8.2.1

The Asterisk Development Team has announced security releases for the following versions of Asterisk: * 1.4.38.1 * 1.4.39.1 * 1.6.1.21 * 1.6.2.15.1 * 1.6.2.16.1 * 1.8.1.2 * 1.8.2.1 These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, 1.8.1.2, and 1.8.2.1

Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve

Hello everybody, I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? Thanks. I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers The one that works: Allow: INVITE, ACK,

Hello, I'm using samba 3.5.11 to connect a Windows 2003 Active Directory. With cups, samba is an part of a print server used to print to windows desktop shared printers. DNS are Active Directory Integrated. Network is both IPV4 and IPV6, IPV6 for Linux and Windows Vista and above. Some times, some users are not able to print. In logs of cups, I see to thinks "Unable to connect CIFS

Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP

Hello, I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via

Hi, I''m working on a french website and I use the Form.serialize method to send the info through AJAX. The thing is that the accentuated letters (é,ê,à, etc.) don''t get replaced by their HTML entities and they get corrupted when retrieving the data. How could I fix that? thanks a lot, Blaise Bernie

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

Hi Everyone, I am currently running Trixbox 2.6 and I have a problem with Asterisk. /etc/init.d/asterisk status Asterisk dead but subsys locked I deleted all files in /var/run/asterisk folder and asterisk restart... It's ok for a while. But some days after Asterisk again is dead. Can anybody help me? Rgs / budacsik

I''m getting more and more "execution expired" as systems checking and hit puppetdb for the first time (switching from a mysql instance). The command queue isn''t long (1-5, if anything, all the time), and ym master itself seems to be dealing well enough. I have seen the collection time growing higher and higher though. This is a ~2K node deployment, and one of the few

Hello everybody, I was wondering what is postponing the 1.4.26 release? I thought it was scheculed for last week. Is there something we can do to help to release this version? There is no more issue reported on https://issues.asterisk.org/ for the time being. Best Regards, -- -- Marc LEURENT lftsy at leurent.eu -------------- next part -------------- An HTML attachment was scrubbed... URL:

so when I''m reading the pick axe book second edition I don''t see anything like the syntax you see people using in rails code. Specifically when you see. belongs_to :Person has_many :Phones etc these are methods on ActiveRecord right? Why is this invocation syntax never described in the Pick Axe book? I do see things like attr_reader :some_attribute etc but you

Hello all, I have a TC400B Digium card in order to deal with transcoding and I have some trouble using it, I have a timer synchronisation problem! I would be very grateful if you have any idea to help me? It seems that the card is not correctly synchronised to the system because when I speak to one side, the sound takes 5 seconds to go to the other side, and increasing, after 30 seconds of call,

Borders show it has to be ordered but Amazon says they have it but don''t show any reviews so what exactly is the true availability of the second edition? --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to

Hey guys, So, we're testing Gluster RDMA storage, and are having some issues. Things are working...just not as we expected them. THere isn't a whole lot in the way, that I've foudn on docs for gluster rdma, aside from basically "install gluster-rdma", create a volume with transport=rdma, and mount w/ transport=rdma.... I've done that...and the IB fabric is known to be

Hi! I want to detect brute-force password hacking attacks - thus if there are too many failed login attempts for a SIP account I want to "lock" this account. Does somebody have any ideas how this could be implemented? thanks klaus