Posts Tagged g.729
AsteriskNOW 1.7.1 now available with add-on installer module for FreePBX
Asterisk NOW
AsteriskNOW 1.7.1 has been released with a new module for FreePBX that allows installation of Digium add-on software from within the web-based interface. Now there’s no command-line work to be done to get Digium’s G729 codec, Fax for Asterisk, HPEC, or Skype for Asterisk. We’ve also made some changes to make AsteriskNOW even friendlier for newcomers to the Asterisk community.
Together with the DAHDI configuration module that began shipping with 1.7.0, these modules make Asterisk administration even easier. Download AsteriskNOW today, and burn it to a CD or start it in a virtual machine. In minutes you can turn your computer into your next phone system!
Asterisk 1.8.0-Beta3 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta3.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.
http://www.asterisk.org/asterisk-versions
This release contains fixes since the last beta release as reported by the community. A sampling of the changes in this release include:
- Fix a regression where HTTP would always be enabled regardless of setting.
(Closes issue #17708. Reported, patched by pabelanger) - ACL errors displayed on screen when using dynamic_exclude_static in sip.conf
(Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson) - Support “channels” in addition to “channel” in chan_dahdi.conf.
(https://reviewboard.asterisk.org/r/804) - Fix parsing error in sip_sipredirect(). The code was written in a way that did a bad job of parsing the port out of a URI. Specifically, it would do badly when dealing with an IPv6 address.
(Closes issue #17661. Reported by oej. Patched by mmichelson) - Fix inband DTMF detection on outgoing ISDN calls.
(Patched by russellb and rmudgett) - Fixes issue with translator frame not getting freed. This issue prevented g729 licenses from being freed up.
(Closes issue #17630. Reported by manvirr. Patched by dvossel) - Fixed IPv6-related SIP parsing bugs and updated documention.
(Reported by oej. Patched by sperreault) - Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a list of a specified item. Matches up with FIELDQTY() and CUT().
(Closes #17713. Reported, patched by gareth. Tested by tilghman)
Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:
- Secure RTP
- IPv6 Support in the SIP Channel
- Connected Party Identification Support
- Calendaring Integration
- A new call logging system, Channel Event Logging (CEL)
- Distributed Device State using Jabber/XMPP PubSub
- Call Completion Supplementary Services support
- Advice of Charge support
- Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta3
Thank you for your continued support of Asterisk!
Digium Releases TCE400B PCI-Express Transcoder Card
Digium TCE400B
Digium releases the TCE400B PCI Express card for use with voice applications based on the open source Asterisk telephony platform. The new card provides hardware-based voice compression and decompression (codec) capabilities to shift transcoding from software to hardware. Using the TCE400B in place of a software-only solution places fewer demands on servers and frees up Asterisk to more efficiently process calls and to provide functionality for phone systems such as call recording, conference calling and interactive voice response.
The TCE400B is a half-length, low-profile PCI-Express x1 card for transforming complex VoIP codecs into simple codecs. This product is, essentially, a PCI-Express version of Digium’s existing TC400B product.
G.729 and G.723.1 codecs x86 and x86_64 Linux and FreeBSD binaries for Asterisk open source PBX
DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm.
Sources
To compile the codecs you need Intel IPP libraries installed. Currently only Asterisk 1.4, 1.6 and TRUNK are supported. Support for Asterisk 1.2 and Callweaver is planned, for now use the binaries. Use “g723 debug” and “g729 debug” commands to print statistics about received frame sizes, can aid in debugging audio problems. You need to bump Asterisk verbosity level to 3 to see the numbers.
- asterisk-g72x-1.0-beta8.tar.bz2
- asterisk-g72x-1.0-beta7.tar.bz2
- asterisk-g72x-1.0-beta6.tar.bz2 – use beta6 for Asterisk 1.6.0.x only, use beta8 for everything else, including 1.6.1
Binaries
- choose codec binary appropriate for your Asterisk version and CPU type, use x86_64 for 64-bit mode, scroll to the end of the list for FreeBSD binaries
- delete old codec_g729/723*.so files (if any) from /usr/lib/asterisk/modules directory
- copy new codec_g729/723*.so files into /usr/lib/asterisk/modules directory
- restart Asterisk
- check the codec is loaded with ‘core show translation recalc 10′ on Asterisk console (‘show translation’ in Asterisk 1.2)
- G.723.1 send rate is configured in Asterisk codecs.conf file (Linux Asterisk 1.2, 1.4, 1.6, TRUNK and Callweaver, FreeBSD 7.x Asterisk 1.4 binaries only):
[g723] ; 6.3Kbps stream, default sendrate=63 ; 5.3Kbps ;sendrate=53
This option is for outgoing voice stream only. It does not affect incoming stream that should be decoded automatically whatever the bitrate is.
- in sip.conf or/and iax.conf configure the codec(s) either globally or under respective peer, for example:
disallow=all allow=g729
- for detailed information about Asterisk configuration visit voip-info.org
- in case of problems read Notes and Troubleshooting
Digium TC400B
Digium TC400B
The TC400B is a half-length, low-profile PCI 2.2-compliant card for transforming complex VoIP codecs into simple codecs.
Overview:
The TC400B is a bundle of the half-length, low-profile PCI-2.2 compliant TC400P base card and the TC400M voice processing module. The TC400B is designed to handle, in dedicated DSP resources, the complex codec translations for highly compressed audio as would otherwise be processed by Asterisk in software.
Asterisk, in software and with Digium G.729a licensing, is capable of transforming the G.729a codec into other codecs for the purposes of call origination or termination, bridging disparate calls, or VoIP to TDM connectivity. These transformations in software are very expensive, in terms of MIPS, and require a substantial amount of CPU time to accomplish. The TC400B not only relieves the CPU of this duty, freeing it up to handle other tasks or to complete additional call processing; but also provides Asterisk with the capability of bridging G.723.1 compressed audio into other formats, a capability not previously possible.
The TC400B decompresses G.729a (8.0kbit) or G.723.1 (5.3kbit) into u-law or a-law; or, compresses u-law or a-law into G.729a (8.0kbit) or G.723.1 (5.3kbit). The TC400B is rated to handle up to 120 bi-directional G.729a transformations or 92 bi-directional G.723.1 transformations. The TC400B does not require additional licensing fees for the use of these codecs nor does it require the registration process associated with Digium’s software-based G.729a codec licensing.
TC400B Documentation:
Manuals
01.26.2007 – TC400B User’s Manual (PDF)
Datasheets & Brochures
07.01.2008 - TC400B Datasheet (PDF)
Agreements & Policies
08.02.2007 – Digium End-User Purchase and License Agreement (HTML)
