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!

Related posts:

1. AsteriskNOW 1.7 Hits The Streets
2. Skype for Asterisk: Public Beta available
3. Fax For Asterisk 1.1.6 Release Announcement

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!

Related posts:

1. Asterisk 1.8.8.0-rc1 Now Available
2. Asterisk 1.8.0-Beta2 Now Available
3. Asterisk 1.8.4-rc2 Now Available

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.

• PCI Express Form Factor
• Half Length, Low Profile
• 120 G.729a or 92 mixed channels of G.729a/G.723.1 codecs

The TCE400B does not provide a timing interface like other Digium telephony card products. As such, should you use the TCE400B with an Asterisk application that requires timing, e.g. MeetMe, then Digium recommends use of either another Digium telephony card or the use of the dahdi_dummy software timing device. Note that the release of DAHDI 2.2.0 brings many improvements to dahdi_dummy.

In addition to its support for G.729a transcoding, the TCE400B gives Asterisk the ability to convert G.723.1 compressed audio into other formats, a capability not otherwise possible with Asterisk or software-only solutions. The card’s capabilities allow transcoding between simple, G.711 u-law and a-law, and complex, G.729a 8.0 kbit/s and G.723.1 5.3/6.3 kbit/s, codecs. When running in G.729a mode, the TCE400B can support 120 simultaneous transformations; in a mixed G.729a and G.723.1 mode, it supports 92 simultaneous transformations.

The TCE400B card product utilizes Digium’s wctc4xxp driver file that is part of the DAHDI driver package. Operation of the TCE400B requires a minimum DAHDI release of 2.0.0. Digium always recommends that customers wanting to use the TCE400B use the current (2.2.0) versions instead of the minimum versions. Current versions may include important bug or performance improvements.
The TCE400B also utilizes the codec_dahdi Asterisk module. Operation of the TCE400B, in addition to the above DAHDI requirements, also requires a minimum Asterisk release of 1.4.22 (or later) or 1.6.0.1 (or later). Digium always recommends that customers use the current versions instead of the minimum versions.
The TCE400B is compatible with Open Source Asterisk 1.4.
The TCE400B is compatible with Open Source Asterisk 1.6.
The TCE400B is compatible with Asterisk Business Edition C.X
The TCE400B is not compatible with any current version of Switchvox SMB.
The TCE400B is not compatible with any current version of Switchvox SOHO.
The TCE400B is not compatible with any current version of Switchvox Free.
The TCE400B’s PCI Identifier is d161:8004

Asterisk is the most widely used open source telephony engine and tool kit. By offering flexibility to access and alter the software code, Asterisk empowers developers and systems integrators to create VoIP communication solutions that match the specific needs of a business. Many businesses use Asterisk as a PBX to manage phone calls, but it’s also commonly used as a gateway between IP and PSTN networks, a telephony feature server and as the basis for call center applications.

Related posts:

1. Digium TC400B
2. Fax For Asterisk 1.1.6 Release Announcement
3. G.729 and G.723.1 codecs x86 and x86_64 Linux and FreeBSD binaries for Asterisk open source PBX

• Sources
• Binaries
• Notes and Troubleshooting
• Getting help

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

Notes

• After download check MD5SUM.
• Use x86_64 build if running 64-bit mode. Binary without x86_64 in its name is 32-bit. Use pentium4/core2/opteron binaries even your processor is 64-bit capable but you are running 32-bit.
• Use Pentium 4 build for Pentium D. Use Pentium 3 for VIA C3 and Pentium 4 for VIA C7. XEON is Pentium3/Pentium4/Core2 with larger cache. Celeron is Pentium3/Pentium4/Core2 with smaller cache. Basically, there are 4 software cores: MMX, SSE, SSE2, SSE3. Pick closest match from Intel CPU line and try it.
• GCC, GCC4 or ICC – all binaries are compatible with Asterisk built by any other compiler, use what is faster or better for you.
• sse3 builds are only for newer P4 chips (Prescott) with SSE3 support, check for PNI flag in /proc/cpuinfo.
• ICC cores are not always the fastest, check with ‘(core) show translation recalc 10′ on Asterisk console, ensure the box is idle or else timings will be inaccurate.
• Non-x86 architecture CPU-s are not supported. IPP is available for Itanium but I don’t have the hardware. There is also IPP for ARM, but I don’t have ARM hardware too. Also, ARM vary a lot, most of ARM-s are too slow to be practical for G.729 and G.723.1, and various things including floating point must be sorted out before ARM binaries became possible.

Troubleshooting

• Noise or metalic sound with codec_g723
  Configure /etc/asterisk/codecs.conf [g723] sendrate=63 or 53 as described at the top of this page.
• Distorted sound
  Disable VAD and CNG in phone and/or your VOIP provider.
• show translation – codec is not loaded
  Bump asterisk verbosity level with -vv and watch for messages right after ‘loading module codec_g729…’
• Module codec_g72… does not provide a description
  Choose proper version of the codec to match your Asterisk version. TRUNK is for SVN TRUNK – the version that eventually became 1.8.
• Module codec_g72… was not compiled with the same compile-time options as this version of Asterisk.
  Try new 1.4 and/or TRUNK builds.
• loader.c… __load_resource: /usr/lib/asterisk/modules/codec_g72…: cannot restore segment prot after reloc: Permission denied
  Execute:
  $ chcon -t textrel_shlib_t /usr/lib/asterisk/modules/codec_g72[39]*.so
  This happens when SELinux is enabled and Asterisk process cannot load the codec because of ELF TEXTREL in shared library. This is how IPP works and that is fine.
• Error loading module codec_g72…: cannot open shared object file: No such file or directory
  Check
  $ ldd codec_g729.so
  $ strace -o trace.log -efile -f asterisk
  send output and log file to Asterisk G.729 Google group.
• Error loading module codec_g72…: wrong ELF class: ELFCLASS32
  Use x86_64 binaries because you’re running 64-bit OS.
• $ ldd codec_g729.so: not a dynamic executable
  You are running 32-bit, see Notes – note #2
• Asterisk crashing at startup
  If Pentium4 codec is not working, try -no-sse builds, try Pentium3. Try ICC builds. Doesn’t work – try Pentium and Debug builds.
• Asterisk still crashing at startup
  Compile asterisk with CFLAGS=-g
  Run asterisk in gdb (with original and -debug version of the codec):
  $ gdb /usr/sbin/asterisk
  (gdb) r -cvvv
  when it crashes do:
  (gdb) set disassembly-flavor intel
  (gdb) disassemble $eip $eip+20
  (gdb) info registers
  (gdb) info threads
  (gdb) bt
  (gdb) l
  (gdb) q
  Send an output to Asterisk G.729 Google group together with the result of ‘cat /proc/cpuinfo’.

Getting help

The primary source of help is Asterisk G.729 Google group. Post your questions there, but first read Notes and Troubleshooting sections above. When reporting a problem it is up to you to provide as much usefull information as possible. “Doesn’t work” – is not a good description.

Related posts:

1. Digium Releases TCE400B PCI-Express Transcoder Card
2. 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)

See articles for this product in our Knowledge Base  >>

Related posts:

1. Digium Releases TCE400B PCI-Express Transcoder Card
2. Fax For Asterisk 1.1.6 Release Announcement
3. AsteriskNOW 1.7.1 now available with add-on installer module for FreePBX