similar to: AST-2019-005: Remote Crash Vulnerability in audio transcoding

The Asterisk Development Team would like to announce security releases for Asterisk 13, 15 and 16. The available releases are released as versions 13.28.1, 15.7.4 and 16.5.1. These releases are available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk/releases The following security vulnerabilities were resolved in these versions: * AST-2019-004: Crash when

Asterisk Project Security Advisory - AST-2009-005 +------------------------------------------------------------------------+ | Product | Asterisk | |---------------------+--------------------------------------------------| | Summary | Remote Crash Vulnerability in SIP channel driver |

Asterisk Project Security Advisory - AST-2009-005 +------------------------------------------------------------------------+ | Product | Asterisk | |---------------------+--------------------------------------------------| | Summary | Remote Crash Vulnerability in SIP channel driver |

There is a serious bug in Sangoma transcoding cards. The card has an internal, small jitter buffer and it drops samples from the audio stream when there is high jitter in the network. The bandwidth is cheap now so for me the only reason to use transcoding is where I have low-bandwidth-high-jitter links. Sangoma said they will not fix it and we had to go back to software transconding. Do you have

The Asterisk Development Team would like to announce the release of Asterisk 13.29.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.29.0 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 16.6.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.6.0 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 13.28.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.28.0 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 16.5.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.5.0 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:

I am unable to get re-invite to work on a new system. Also, unwanted transcoding is occurring on PSTN calls. The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2, CentOS 5.8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on

Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I

Hi there, up till now I had this two-box setup in mind: * no.1: public IP * no.2: private IP, registers with no.1, serves a small office with clients behind NAT See we'd get something like this: SIP client (GSM) --> *1 --> IAX2 (iLBC) --> *2 --> G.711 --> MGCP UA The codec of the SIP client (on the Internet) I don't have full control over, that depends on the

Hi all, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice

Forwarding a conversation with a company that provides a management tool for transcoding Vorbis files to MP3 when communicating with the Nomad Jukeboxes from Creative. Good initiative, and probably sounds OK too, but would be even better with FLAC, so I'll ask about it. Linus ---------------------------- Original Message ---------------------------- Subject: Re: Vorbis question From:

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963] zttranscode: Loaded. [ 7.618969] wctc4xxp: tc400b0: Attached to

Asterisk Project Security Advisory - AST-2013-005 Product Asterisk Summary Remote Crash when Invalid SDP is sent in SIP Request Nature of Advisory Remote Crash Susceptibility Remote Unauthenticated Sessions Severity Major

Asterisk Project Security Advisory - AST-2013-005 Product Asterisk Summary Remote Crash when Invalid SDP is sent in SIP Request Nature of Advisory Remote Crash Susceptibility Remote Unauthenticated Sessions Severity Major

The Asterisk Development Team would like to announce the release of Asterisk 17.1.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.1.0 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:

As far as I can see, transcoding could be usefull for people who do not primarly care about quality but about filesizes. One could assume that such a user would have a collection of mp3's at 128kbps or higher bitrates, and uses an encoder like BladeEnc or Xing. He wants to take uses of ogg's supposed quality and transcode his 128-or-higher files into 96 or 112kbps oggs to save diskspace.

Hello I use asterisk realtime, and I can set the order of codec preference on my realtime allow column. If I could disable transcoding, then I can always ensure a passthrough of the common codec from origin to destination without transcoding (expensive on CPU) - and more or less, force the codec to use by setting the codec preference So, can I disable transcoding?

Asterisk Project Security Advisory - AST-2014-005 Product Asterisk Summary Remote Crash in PJSIP Channel Driver's Publish/Subscribe Framework Nature of Advisory Denial of Service Susceptibility