[Asterisk-Users] Re: www.openpbx.org
On Friday 07 October 2005 19:10, Troy Settle wrote: > Nice smartass remark... of course anyone can register a domain name. > > Is forking asterisk legal? Of course it is! Asterisk is under the GPL, > which means that anyone can fork it at any time for any reason. > > Look at this in a positive light... many open source projects have > forked, and the branches almost always end up feeding on one another. The difference in this case being of course that OpenPBX can happily continue to feed on any good developments (code wise) that happens in Asterisk but due to Digium's dual license restrictions Asterisk will not be able to feed on code that goes into OpenPBX. This means (and this is already the case if you look at the source tree) that other issues aside, OpenPBX should progress quicker in the long term. (That is if you assume that there is good code in Asterisk that the OpenPBX developers wish to use and visa versa) Cheers -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: OPAL now supports IAX2
On Sunday 07 August 2005 04:05, Brian West wrote: > > What are the advantages of using woomera IAX2 instead of native IAX2? > > Put woomera aside right now, This is something that brings a cross > platform IAX2 stack that can for example be used in Gnomemeeting or > anything else that uses OPAL, using a closed and open familiar API. > This can be used on windows, linux and anything that OPAL and PWLIB > can be used on without any changes. Its a step in the right > direction in my opinion. To clarify things a little, OPAL is a cross platform (Unix, Linux, Mac and Windows) telephony library written in C++ which among other things supports H323, SIP and now IAX2 thanks to Derek Smithies. It (and its older brother OpenH323) allows you to use a large number of Open Source Audio (including wideband) and Video codecs as well as commercial ones (like g729) as it is distributed under the MPL license. The design of the API allows things like codec translation (GSM-g711) applications or protocol translation (SIP-H323-IAX2) applications to be written in 20-30 lines of code. (Video codec translation is also possible of course) These and many other features can be used from client applications (SoftPhones, IP Phones, Gateways) and from servers like Asterisk. Thanks to woomera (another app from Craig Southeren) and chan_woomera (an Asterisk channel by Anthony Minessale) the VoIP Signalling and Media handling (RTP or IAX2) can be offloaded onto OPAL or OpenH323, thus bypassing many of the stability issues caused by the poor locking and thread model in Asterisk. Of cause an added bonus is that you may use commercial/patented/closed source VoIP codecs without voilating Asterisk'S GPL license. A second implementation of the IAX2 protocol also helps with finding bugs (and other problems) in the reference implementation in Asterisk. Although the IAX2 code in OPAL is still very new, the heavily multithread object nature of OPAL should allow much higher scalability than the existing library allows. I hope that helps explain some of the possibilities -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Chan_Woomera beta released at www.pbxfreeware.org
On Thursday 23 June 2005 22:41, Brian West wrote: > chan_woomera is another alternative h323 implementation. > > visit www.pbxfreeware.org for more information. > > Thanks, > Brian > www.cluecon.com On behalf of everyone I would like to thank Brian, Anthony and Craig for their hard work. We have all been waiting for this... -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
On Monday 30 May 2005 13:28, Matteo Brancaleoni wrote: > and , what is more interesting, > they've omitted any reference to digium resellers > and specified only distributors :( Yes. Our reseller info was removed. And some of our customers have been sold to directly.. Not a nice way to do business :-( -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
On Monday 06 June 2005 02:50, Kevin P. Fleming wrote: --snip-- > > I've just not been impressed with Digium's behaviour lately. They've > > gotten quite hostile over Sangoma hardware lately, claiming that Sangoma > > (by continuing to develop, refine, and expand their hardware lines, > > which are much older than asterisk, and which asterisk was originally > > developed on) are just ripping them off. If anything, Digium is ripping > > people off with hardware which is inferior (though I've seen claims that > > they have some good new stuff coming - excellent!). > > Exactly where did you hear that Asterisk was "originally developed on > Sangoma hardware"? Nothing could be farther from the truth; the very > earliest Asterisk code supported some funky Voice-over-Frame-Relay > hardware, then it was modified to support the Zapata hardware from the > Zapata open source hardware group. Digium has never had any drivers for > Sangoma hardware, and still doesn't. Any Asterisk support for Sangoma > hardware has been built by Sangoma. --snip-- "When I wrote Asterisk originally, I used a Sangoma Frame-Relay card to talk to an Adtran Atlas, and used the Atlas to drive a channel bank. I got all this stuff running but it was a pretty big mess at the time," recalls Spencer. "I faced a number of problems in the development of this. One of the primary problems was that of finding hardware to connect the PC to the phone system. The hardware available was as expensive as a PBX itself. Of course, once I got it working, I could add expensive features in software for practically no cost -- other than time to write the code -- but the hardware issue was definitely a problem." Source: http://linuxdevices.com/articles/AT8678310302.html Lets get our facts straight shall we. It appears that the FIRST ever hardware suported by Asterisk was Sangoma ;-) Cheers -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Strategy for a stable IAXy
On Thursday 10 February 2005 20:35, Colin Anderson wrote: > > Why would someone choose these over other boxes, such as the Sipura 2000 > > and 3000? > > Because I want NAT traversal and a low bandwidth codec. That's the whole > point of IAX2 as opposed to SIP. There is no low bandwidth codec available with the IAXy that I know of... Minimum is 32k + IP overhead SIP does work through NAT btw.. -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk and Pleiades P32mxi [followup]
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 15 April 2004 00:38, Peter Nixon wrote: > Hi Guys > > I have an Asterisk box with an E100P card connected to a Pleiades P32mxi > (http://www.pleiadescom.com/p32mxi.html) > > When I set the channel bank and asterisk to use Loop Start (or Kewl Start) > to communicate calls can happily go from asterisk, via the channel bank to > PSTN. However, Asterisk sees these calls as being answered immediately > regardless of if the call is still ringing or actually answered. The > channel bank has Answer Supervision installed (Voice Activity Detection) > and the debug messages on the channel bank show that this is working > correctly. > > It appears that Loop Start signaling does not contain enough information > for Asterisk to detect the ringing state of the call, and at the > recommendation of Pleiades I would like to change my signaling to "E & M" > or "E & M Wink" (Actually they recommend to use "R2 Digital" signaling but > I dont think asterisk supports this). > > The problem is when I configure E & M (Or E & M Wink) on both the channel > bank and asterisk the channel bank fails to see ANY signaling comming over > the E1 interface. Nada, nothing, zilch.. I have spent MANY hours on this > problem with Pleiades tech support, and quite a few calls to Digium also > but I cannot get this working. Does anyone else have a Pleiades channel > bank?? Has anyone else gotten one working with E & M? In a followup to this, so that the list archives will have an answer, and to answer the people who have mailed me privately with the same problem. This problem was due to a bug in Asterisk. Mark has kindly fixed the problem, and CVS dated 2004-04-19 or later should have the fix. The problem is that E&M signaling on E1 trunks actually uses different characters for the signaling to E&M on T1 trunks. Asterisk did not know about E&M E1 signaling so therefore any E1 devices connected to Asterisk via E1 using E&M signaling would not understand what Asterisk was saying The new code in CVS adds a "signalling=em_e1" type for use in zapata.conf which works in the same manner as "signalling=em" but for E1 trunks. Mark has not yet added support for E&M Wink on E1 but hopefully he will do so in the future. Regards - -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD4DBQFAhQnHAcdsUt9pJjwRAp57AJYuhfm6f1zRyCEAoNWRESXCIbKXAKCBiCNC WPtXrWKuVf1PwT1wTmphnQ== =+UpN -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Pleiades P32mxi
Hi Guys I have an Asterisk box with an E100P card connected to a Pleiades P32mxi (http://www.pleiadescom.com/p32mxi.html) When I set the channel bank and asterisk to use Loop Start (or Kewl Start) to communicate calls can happily go from asterisk, via the channel bank to PSTN. However, Asterisk sees these calls as being answered immediately regardless of if the call is still ringing or actually answered. The channel bank has Answer Supervision installed (Voice Activity Detection) and the debug messages on the channel bank show that this is working correctly. It appears that Loop Start signaling does not contain enough information for Asterisk to detect the ringing state of the call, and at the recommendation of Pleiades I would like to change my signaling to "E & M" or "E & M Wink" (Actually they recommend to use "R2 Digital" signaling but I dont think asterisk supports this). The problem is when I configure E & M (Or E & M Wink) on both the channel bank and asterisk the channel bank fails to see ANY signaling comming over the E1 interface. Nada, nothing, zilch.. I have spent MANY hours on this problem with Pleiades tech support, and quite a few calls to Digium also but I cannot get this working. Does anyone else have a Pleiades channel bank?? Has anyone else gotten one working with E & M? TIA -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
