similar to: Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1, and 1.6.2.0-beta4 Release Announcement

The Asterisk Development Team is pleased to announce the the second release candidate of 1.6.0.11, the release of 1.6.1.2, the first release candidate of 1.6.1.3, and the fourth beta of 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ . The release of 1.6.1.2 fixes a remote crash security vulnerability in the RTP stack. The

MixMonitor seems to work: -- User hit '*3' to record call. filename: auto-1250792853-24-22 == Begin MixMonitor Recording SIP/snom2-084c4ec8 /var/spool/asterisk/monitor/auto-1250792853-24-22.wav exists now. Recording a call without mixing fails. > User hit '*1' to record call. filename: wav,auto-1250793354-24-22,m TOUCH_MONITOR_OUTPUT is set to

I'm trying to figure out which 1.6 releases have the fix I'm looking for by reading the ChangeLog for each release, whether it's in the 1.6.0, 1.6.1 branches, or an -rc release. If I look at the latest -rc releases of 1.6.0 and 1.6.1 (which are 1.6.0.11-rc2 and 1.6.1.3-rc1 respectively), will that be an exhaustive list of changes or not? The reason being I'm still waiting on the

Hello, I hope you won't mind if I give you mine opinion and suggestions about Vorbis audio encoder, especially new version. I was pretty satisfied with the quality of Beta4 encoder. For a bitrates around 112 to 128 kb/s I consider it better then any other audio compression formats (MP3, AAC, ...) at same bitrate. Sound quality that this version produces with bitrates below 112 kb/s was not

Hi! I would like to share with you an interesting test result. I re-encoded a song 10 times (the 2. made from the 1., the 3. made from the 2.,... the 10. made from the 9.), with a beta4 (CVS 20010620) and with an RC2 (CVS 20010817) library version, 256 kbit/s mode, channel coupling disabled at RC2. The beta4 is very good at the 10. encoding too, but the RC2 has some interesting quality bugs. So,

/> Additionally, it has been stated many times that you / /> _should_ _never_ _transcode_ _any_ _lossy_ _codec_. / This isn't transcoding. The point about transcoding, say, MP3->OGG is that the two codecs use different assumptions (psychoacoustic models, etc.) in performing the compression -- so transcoding often results in one codec trying to chase down and reproduce artifacts

The Asterisk.org development team is pleased to announced the release of Asterisk release candidates 1.6.0.7-rc2, 1.6.1.0-rc3, and beta release 1.6.2.0-beta1. Additionally, new release candidates of Asterisk-Addons 1.6.0.2-rc1 and 1.6.1.0-rc3 have been created. Note that the 1.6.1 series of Asterisk-Addons is compatible with both Asterisk 1.6.1 and 1.6.2 branches. These releases are available for

After doing an informal (128k) listening test, I have concluded that I prefer Beta4 over RC2. The 16kHz low-pass on the RC2 encoder makes it sound like FM radio. Both encoders SEEM to have a couple of dB bump at 10kHz. JT --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.14 and 1.6.1.5. Asterisk 1.6.0.14 and 1.6.1.5 are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 1.6.0.14 is the first full, non-security release since 1.6.0.10. The release candidate 1.6.0.11-rc1 was redone as 1.6.0.14-rc1 (which this release has been created

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.14 and 1.6.1.5. Asterisk 1.6.0.14 and 1.6.1.5 are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 1.6.0.14 is the first full, non-security release since 1.6.0.10. The release candidate 1.6.0.11-rc1 was redone as 1.6.0.14-rc1 (which this release has been created

Hello, all. After reading the README, UPGRADE.txt, and a quick tour through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, one simply compiles and installs over the old installation being careful to NOT install the sample files? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsullivan at opensourcedevel.com

Hello, So much keeps changing with the dialplan and Realtime lookups. Just downloaded the latest stable 1.6.1.2. The app_realtime, which was perfectly brilliant and did exactly what I needed, is gone; replaced with func_realtime. The REALTIME function is unacceptable: ; Get the conference number from the user exten => s,n(readconfno),Read(USER_CONFNO,conf-getconfno,0,3,20) ; See if

The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.0 is the first feature release since Asterisk 1.6.1.0, which was released April 27, 2009. Many new features have been included in this release. For a

The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.0 is the first feature release since Asterisk 1.6.1.0, which was released April 27, 2009. Many new features have been included in this release. For a

Hi! Been trying to figure out how to stream metadata. I got metadata streaming working for MP3, but just cant seem to make it work for Ogg? This is how I *think* it works - please correct me where I'm wrong: 1. For MP3 the metadata is updated via a HTTP GET (aka Old shoutcast method, just new URL /admin/) 2. For Ogg the metadata is embedded directly into the stream using a comment packet? I

The Asterisk Development Team is pleased to announce the releases of 1.2.34, 1.4.26.1, 1.6.0.12, and 1.6.1.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of 1.6.1.4 fixes a remote crash security vulnerability in the SIP stack. Although this crash was not demonstrated in any other version, the details of the

Hi all, We encountered a strange phenomenon when trying to upgrade from 1.6.2.0 to any newer releases: We use the following cli command to feed a wave/mp3 file into an existing conference on an other serve: /opt/asterisk/sbin/asterisk -r -x "channel originate Local/ConfGongAdmin at XY_Features extension ConfGongPlay at XY_Features" The corresponding extensions.conf part looks like

hello, I tried to compil asterisk-addons-1.6.2.0-rc1, and I have that error: [CC] res_config_mysql.c -> res_config_mysql.o res_config_mysql.c:1367: error: unknown field ?update2_func? specified in initializer res_config_mysql.c: In function ?parse_config?: res_config_mysql.c:1432: error: ?CONFIG_STATUS_FILEMISSING? undeclared (first use in this function) res_config_mysql.c:1432: error:

I have icecast2 running on several machines now and have gotten relaying to work just fine. I've posted my instructions on how to do this to my icecast docs site. I have just a few more questions. Perhaps some one on this list has already addressed these issues but has simply not posted these elsewhere. 1) I'm looking for a startup script for launching the icecast2 icecast ices binaries

http://dovecot.org/releases/1.2/beta/dovecot-1.2.beta4.tar.gz http://dovecot.org/releases/1.2/beta/dovecot-1.2.beta4.tar.gz.sig This release fixes a memory corruption bug related to MODSEQ handling in earlier v1.2 releases. Memory corruption means that it's a security bug and possibly exploitable. MODSEQ code is new in v1.2, so v1.1 and older releases are not affected. Other than that, there