similar to: AST-2017-012: Remote Crash Vulnerability in RTCP Stack

The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available security releases are released as versions 13.13-cert9, 13.18.4, 14.7.4 and 15.1.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolves the following security

Asterisk Project Security Advisory - AST-2017-008 Product Asterisk Summary RTP/RTCP information leak Nature of Advisory Unauthorized data disclosure Susceptibility Remote Unauthenticated Sessions Severity Critical

The Asterisk Development Team would like to announce the release of Asterisk 13.19.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.19.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 15.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.19.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.19.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 15.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Asterisk Project Security Advisory - AST-2017-005 Product Asterisk Summary Media takeover in RTP stack Nature of Advisory Unauthorized data disclosure Susceptibility Remote Unauthenticated Sessions Severity Critical

Asterisk Project Security Advisory - AST-2014-019 Product Asterisk Summary Remote Crash Vulnerability in WebSocket Server Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Moderate

Asterisk Project Security Advisory - AST-2014-019 Product Asterisk Summary Remote Crash Vulnerability in WebSocket Server Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Moderate

Asterisk Project Security Advisory - AST-2008-012 +------------------------------------------------------------------------+ | Product | Asterisk | |----------------------+-------------------------------------------------| | Summary | Remote crash vulnerability in IAX2 |

Asterisk Project Security Advisory - AST-2008-012 +------------------------------------------------------------------------+ | Product | Asterisk | |----------------------+-------------------------------------------------| | Summary | Remote crash vulnerability in IAX2 |

Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machine....When running asterisk -rvvvvv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891

Hi, I am plying with * for 10 days now. I am testing with a couple of vocaltec h.323 gateways (FXO and PRI) cisco ata-186 (configured for SIP) and MSN messenger (SIP). They all seem to interoperate. However I have a problem when * is sending calls to the vocaltec gateways. Vocaltec gateways are monitoring the RTCP reports send from the remote gateway (in this case *) and if they don't get a

This article below came up on the newwire. The RTCP XR RFC was published. Will Asterisk be supporting this function in a future release? Does anyone know if any phone vendors are going to be supporting it? Thanks Lee Goodman Our Technology Update this week is about one of those mechanisms. Known as RTP Control Protocol Reporting Extensions (RTCP XR), the technology defines a standard way to

X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more information X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-From: yusuf at ecntelecoms.com X-Spam-Status: No My third try, humph! Yusuf wrote: > Hi, > > I am trying to understand the RTCP stats in Asterisk. > > 1. I am using the 'h' exten to store the RTCP records in

Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?: ? -- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack ??? --

[Apr 15 11:12:19] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR transmission error to aaa.bbb.ccc.ddd:37259, rtcp halted Operation not permitted [Apr 15 11:12:23] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR transmission error to aaa.bbb.ccc.ddd:38563, rtcp halted Operation not permitted What is the specific nature of this traffic? Despite the above the call still functions. What

http://bugzilla.netfilter.org/show_bug.cgi?id=757 Summary: SIP connection helper not setting RTCP conntrack expectation Product: netfilter/iptables Version: linux-2.6.x Platform: i386 OS/Version: Ubuntu Status: NEW Severity: normal Priority: P5 Component: ip_conntrack

Hi, While looking for the cause of disturbance in call I found this error coming in console RTCP SR transmission error, rtcp halted Google search only shows some bug reports relating to MOH and Hold. What could cause this message? Could this be a symptom causing call disturbance? Where should I start digging to find out the reason for this error? I am using Asterisk 1.4.19 with zaptel 1.4.9.2

Hello list ! SETUP : Grandstream --sip--> Local Asterisk (NSLU) --iax--> Hosted Asterisk (VirtualDedicatedServer) --sip--> SIPprovider --> my CellPhone PROBLEM : I've noticed that when I put down the horn of my Grandstream it still takes a while for my GSM/CellPhone to stop ringing. INFORMATION : This is the output on the CLI of the local Asterisk-server : [Oct 3 17:40:33]