search for: readytechnolog

...hoping that means it's safe to release into the real world. This code has also been released on the -dev list. As it is still somewhat new, I would invite anyone with feedback to forward it to me personally or to the -dev list, as I don't monitor the -users list very often. http://www.readytechnology.co.uk/open/g729

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

Hello, For education purpose, I would like to add G729 support to an Asterisk 1.4.19.1 system. I've read this : http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ but when downloading Intel IPP software, it appears that latest IPP version is 5.3. As installation instructions here ( http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/doc-svn8.txt) relates to Intel IPP 4.2, I thought I should try something else. I...

Hello, is there an open-source implementation of G.729 codec for use outside of US? I know it's a patented codec, but since there are usually no software patents outside of the US, I don't care about the patent license. I could use open-source implementation of the codec, if there was some. Any ideas? Sincerely, Juraj Bednar.

A new release of the open source G.729 patch has been issued. The new URL is: http://www.readytechnology.co.uk/open/ipp-codecs The memory leak in codec_g729 is now fixed. This was due to a problem in a section of code copied from the Intel example. Thanks to those who assisted in locating this bug. If you are still running the old version of the codec, your Asterisk process will run out of memory...

Hello To cut down on the number of hackers trying to break into an Asterisk server, I'd like to simply move the SIP port from the standard UDP 5060 to something non-standard. Since this server must be able to receive INVITEs from any SIP UA (server or client), it appears that I must add an SRV record in the DNS so that they can locate the server and the port used to reach it. _sip._udp SRV

It took a little longer then expected, but here it finally is, a field test for the idefisk for linux iax2 softphone. Freely downloadable from http://www.asteriskguru.com/tools/ You will probably need to copy the iaxclient lib into your library directory and run ldconfig before starting the phone. Please note that this is the first copy in the wild of the linux version and is not as tested

I was somewhat concerned reading Mark's posting earlier today. Obviously, things are very bad in the US at the moment. Their Government even deported Cat Stevens the other day (check http://news.bbc.co.uk/1/hi/england/london/3686992.stm ). Clearly, given the fact that Digium contributes so much to Asterisk, they shouldn't be forced to risk their company's future by hosting these

...mercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <478C7665.1000404 at blueface.ie> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Steve Totaro wrote: > >> >> I would suggest building it yourself >> (http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt >> <http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt>). It is >> not that difficult and ensures that it "should" be compatible with your >> machine. Just a little work. >> > > Has anyone tried building...

I have been impressed with the quality and meagre bandwidth of the G729 codec from Digium. I am in a testing phase of our roll out, we are using 5 Asterisk PBXs in various countries to provide connectivity for our employees, owners and family. As we are testing, and our setup is somewhat complex due to the peculiarities of our connectivity, there has had to be a lot of changes to servers, cards to

Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? Regards, Liangliang

>I'm interested in the g729 diff you posted... > >I've applied the patch, but I don't seem to have the prerequisites to >compile it... I tried downloading the other code available from >Intel, but even the 'eval' version won't install without a FlexLM >license (damn license managers...). Am I heading the right direction, >or is there somewhere else I

-- snip -- Had the patch been against the actual g729 libraries the case would have been clear. Now, the patch is against asterisk to make it interoperate with the g729 libarary and this may or may not be non-infringing. However, the distribution of the g729 libraries themselves are almost certainly infringing. There is also the possibility that the patch to asterisk may be ruled a

For those of you coming to FOSDEM on 24/25 Feb, there'll be a session in the Debian devroom on Open Source VoIP. http://www.fosdem.org/2007/schedule/speakers/daniel+pocock Several VoIP projects will be represented in various ways throughout the weekend, and there will be some of the following: - hardware giveaways from leading VoIP companies - launch of new open source VoIP product

Hi all, All the examples I've come across seem to suggest configuring jabber.conf/jingle.conf/gtalk.conf for a real Google account. What about the scenario where the Asterisk server should connect to an account on a private Jabber server and using Jingle (voice calling over Jabber)? e.g. for the domain widgets.com: - there is a copy of ejabberd running on the same box as Asterisk, and

I've raised a bug report about this here: https://issues.asterisk.org/jira/browse/ASTERISK-19268 I'm just wondering who else has been investigating RFC 5922 style certificate practices? Which CAs have been able to provide appropriate certificates? What kind of interoperability testing has been done between the major products (e.g. Asterisk, Kamailio, OpenSIPS, reSIProcate/repro)?

I've just come across this issue: https://issues.asterisk.org/jira/browse/ASTERISK-17727 I am strongly in support of TLS and I believe this issue will be a stumbling block for more and more users - because more and more CAs are using the intermediate certificate chains For example, the free startssl.com certs are trusted by Android phones now. I have a UA running on my phone against a SIP