similar to: SIP or IAX

We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara

Hey, I am here at Astricon and about to go down to registration. Is there any interest in pictures if I take my digital camera? I am sure that someone is already doing this. (Probably someone official). I would take pictures of each day and upload them to my website if anyone is interested. Let me know! -- Kristian Kielhofner

Friends, Please send all offers for help *off list* to us at info@astricon.net. Do not disturb the list with offers of your services, please. I repeat: Only the Developer's Meeting will be considered for broadcast at this time. In order to enjoy the conference, you will simply have to be there. It's an IRL experience - meeting all the other Asterisk user's from around the globe,

Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID calls, originating and transferring. A provider offers both SIP and IAX trunking. Cateris paribus, what is the preferred solution to choose? What points to consider? I can name the provider if this is not against this list policy--is it? Thanks, -kkm

Astricon 2004 is the first Asterisk user's and developer's conference, to be held in Atlanta, Georgia in September. See http://www.astricon.net ** We will soon open for early bird registrations! ** To get a very low price, I recommend that you participate as a speaker. In fact, speakers doesn't pay any fee at all to participate in Astricon. **** Send us your speaker's proposal

I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be

We're now opening up registrations for the Astricon tutorials again. We've been able to move to new conference rooms within the same hotel. Register on line at http://www.astricon.net We're sorry for the inconvienience our recent closing of the tutorials may have caused you. You are welcome to contact us at info@astricon.net to change your reservation to include the tutorials. There

I know this is probably way out there but............ Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an

Hi guys, I am following this[1] tutorial to install clang. However, I have made a small change in the configure command, and I am running it with '--enable-optimized' option to avoid the debug build. I am getting the errors (given at the end) related to *undeclared identifiers '__NR_*'*. Can someone please provide some input about how to tackle this issue? On my other machine, I

Thank you very much for all feedback on Asterisk Sunday News! This is the last issue for June. This week I'll go on holiday and will be back with more news in early July. My kids are getting summer leave this week and we'll be visiting the south of England for a while. Another part of Europe that still use their own currency. If you think there's an European standard, you're

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hmm, My music on hold has always worked fine. But I discovered that under incoming IAX2 calls they don't get any MOH! All I could find was a comment saying let me know if you find a solution... Nor does the debugger does say: Started music on hold So it's not starting the MOH, why? I do have it configured and it does play under other

Hello Asterisk users around the world! Recently, I have been working with pretty large Asterisk installations. 300 servers running Asterisk and Kamailio (OpenSER). Replacing large Nortel systems with just a few tiny boxes and other interesting solutions. Testing has been a large part of these projects. How much can we put into one Asterisk box? Calls per euro invested matters. So far,

Pardon the newbie post. I installed Asterisk on a "test" system using the Asterisk@home cd image. When you boot from Asterisk@home is installs an O/S and Asterisk on your PC. How cool is that. But I was wondering if someone could point me in the right direction for updating the version that I have. I'm new to CVS, how do I determine what version to build? Is there a primer on how to

Dear Asterisk Community, Asterisk 1.0 was released at Astricon 2004, in September last year. It's been almost a year and we haven't been able to go ahead and release a new version. Now is the time to try to move forward again. As we've outlined before, the process is this: -------------------------------------------------------------------- * Code freeze: At this point, we'll

How do you configure an outbound proxy for Asterisk? If the sip call is not local I want everything to go to a designated sip proxy. Thanks, Chad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040913/bdd57a91/attachment.htm

hey all, am wondering if anyone has successfuly done a SIP attended transfer using the REFER method (after an INVITE obviously) and the Replaces: header. we're writing our own SIP UAC and the asterisk code seems to support it, but we're not really sure if this is so. we plan on the following call flows: 1. incoming call from exten 1111 is sent to our UAC with Dial() 2. our UAC makes

I am wondering if there are any tricks getting the Cisco ATAs to do "distinctive rings" via the ALERT_INFO variable? I have seen some contradictory information in the Wiki, and I tried the example there. I then sniffed the connection between the server and the ATA and didn't see the header sent like it is "supposed" to be. If someone out there has a handle on this and

I have to do Load balancing and Qos on a Linux box which is having openvpn and iptables running System is Fedora core 1 Any known issues ? Or anything I have to take care ? Thanks -- Sandeep A.S <sandeep@netcontinuum.com> Netcontinuum Pvt Ltd _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc

Remote work possible. Location: Columbia, SC Duration: 12+ months Rate: $60/hr Daily Responsibilities: Ruby coding, using rails, testing, generating ad hoc reports for client as needed, providing daily status updates to team, Presence on internal IM system during work hours. Required Skills : RUBY, RAILS, GIT, MYSQL, CUCUMBER, RSPEC, JQUERY, EXCELLENT, LINUX, OS X, JSON, COMMAND

On 11/24/16 3:12 AM, Arnaud De Grandmaison via llvm-dev wrote: > > Hi Sandeep, > > The CMake version of the test-suite has been improved (see > https://reviews.llvm.org/D21360) so that it’s easy to plug additional > test suites. A starting point would be to look at how the > TEST_SUITE_SUBDIRS variable in the top level CMakeLists.txt is used. > I think this bug report