2004 July

July 2004


\"AsteriskAnother issue of Asterisk Summer News, delivered right to your
mailbox! Back here in Sweden, it\’s finally summer weather.
Sunshine and some heat. It\’s good for our ice bears and
the snow houses to get some sunshine :-)

Asterisk development and IRC chat has gone into a lazy summer
mode, but the mailing list is still cooking. It\’s impossible
to keep up with it, for both gurus and newbies, even during
summer holidays.

This issue will be a short issue with just a few articles.
Enjoy!

This week\’s topics:

  • Asterisk 1.0rc1: Feedback, please
  • Astricon 2004: Early bird discount only applies in July
  • Asterisk IRC chatters: BEHAVE!
  • Open Source VoIP Watch: SER 0.8.14
  • Dialplan updates: The DIAL() application
  • Recent CVS changes

(more…)

Linux Magazine publishes an in-depth article by Thorsten Späth about Asterisk in their August 2004 issue. It\’s a great introduction to Asterisk for new users and gives a lot of advice on how to install and configure Asterisk.

\”Asterisk is one of the most powerful freeware
telephone systems, period. The
phone system function is just one of
Asterisks features. You can use Asterisk
as a VoIP gateway, an exchange, or even
for providing telephony services. It connects
the world of traditional phone
systems with the more modern world of
TCP/IP. Besides a steadily increasing user
base and improved hardware support,
Asterisk is slowly but surely becoming a
one-stop communication hub.\”

This RFC updates the S/MIME requirements in RFC 3261, making it a bit easier to implement both TLS and S/MIME. Before this, there was a need to implement both 3DES for S/MIME and AES for TLS in a SIP phone, now it\’s only AES.

Unifying the ciphersuite and signature algorithm requirements for TLS and S/MIME would simplify security implementations. It is therefore desirable to bring the S/MIME requirement for SIP into parity with ongoing work on the S/MIME standard, as well as to unify the algorithm requirements for TLS and S/MIME. To date, S/MIME has not yet seen widespread deployment in SIP user agents, and therefore the minimum ciphersuite for S/MIME could be updated without obsoleting any substantial deployments of S/MIME for SIP (in fact, these changes will probably make support for S/MIME easier).

After a lot of work being done to get Asterisk to run smoothly on FreeBSD, the port to Mac OS X is now also running. You can use Asterisk on Mac as a one-person PBX or on a server as an office PBX.

People have been able to get Asterisk to compile and run on MacOS X (cache) systems, including MacOS X \”Jaguar\” (ver 10.2) and \”Panther\” (ver 10.3). There are routines in the standard makefile to accomodate MacOS X, whose open-source subsystem Darwin (cache) is based on FreeBSD (cache) and the Mach 3.0 (cache) microkernel.

Techdirt published an article about Nigerian natives that believes some phone numbers are lethal. Maybe we should invent a class of SIP URI:s that are magical? I don\’t know if this is just a myth invented during a lack-of-news period or actual facts, but if it has got any truth in it, it\’s a culture clash worth noting:

An urban legend is making its way around Nigeria that has many people afraid to answer their phones. Apparently, a large number of people actually believe that picking up phone calls from specific numbers will kill them immediately. A list of phone numbers is being circulated, and it must suck if you actually have one of those phone numbers. The mobile phone operators in the country have been trying to get the message out that this is untrue, but apparently it\’s not traveling too well (one imagines the message is not being passed along via phone calls).

\"AsteriskThis week starts with the exciting news: We\’re getting close to
Asterisk 1.0 again. After the failed attempt earlier this year,
we\’ve been able to remove a lot of the MAJOR/CRASH bugs from the
bug tracker and Mark feel\’s it\’s time to target 1.0 again.

At this point, the community needs to work as a community,
spending extra time on finding bugs, solving issues, improving
documentation and making Asterisk more stable. There has been
extensive code reviews, but the more eyes that go through the
source, the better. Telephony requires stability.

Join the effort, regardless if you are a user,
administrator, coder or documentation writer!

Consider this: You\’re drafted! :-)

This week\’s topics:

  • Asterisk 1.0: Second try
  • Astricon 2004: Early bird discount only applies in July
  • Changes in the #asterisk IRC channel – Registration required
  • Asterisk Developer of the week
  • Asterisk GUI of the week
  • Using Call parking with CVS head? Rename the config file!
  • Sunday News Re-run: Read the configs, Luke (even if you\’re an Asterisk guru)
  • Chan_sip2: Now updated
  • Recent CVS changes
  • Reporting bugs in the bug tracker

(more…)

On Saturday, Mark Spencer release the first release candidate for Asterisk 1.0. A previous attempt at reaching 1.0 failed, since there was too many serious bug reports that required massive changes to the Asterisk source code. At the same time, bug marshals implemented a feature freeze, postponing the addition of several patches in the bug tracker to after the 1.0 release.
Mark wrote:

We have officially made the first release candidate of Zaptel, Libpri, Asterisk and Gastman available. While there are still open major bugs, they are relatively limited, and it was time to go ahead and get the 1.0 ball rolling in earnest.

Free World Dialup® has passed the 250,000 member mark! FWD® continues to be one of the largest worldwide IP Communications networks built on open standards….

…congratulations to Jeff and Ed! The Free World Dialup network is a great and innovative platform for the VoIP revolution!

It was a matter of time before this got out to the press. Caller IDs from VoIP customers should not be trusted when moving into the PSTN. And we should not send secret phone numbers out to customers.

Hackers have discovered that implementation quirks in Voice over IP make it easy to spoof Caller ID, and to unmask blocked numbers. They can make their phone calls appear to be from any number they want, and even pierce the veil of Caller ID blocking to unmask an anonymous phoner\’s unlisted number.

David Isenberg wrote an interesting article about SIP and IAX, Asterisk\’s VoIP protocol, in America\’s Network, June 15:

I asked Mark Spencer, the author of IAX, another fast-gaining communications protocol that is shaping up as SIP\’s rival, about differences between SIP and IAX. \”SIP violates protocol layering principles,\”says Spencer. In addition, he says, SIP is hobbled by complexity of \”seemingly limitless extensions.\”

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