For the record, Douglas is correct on this point of "enterprise-grade" being on ABE: http://www.digium.com/index.php?menu=product_category&category=software
Copied and pasted right from the website, it says: Asterisk Business Edition(tm) Digium(tm), the leader in open source telephony, offers Asterisk Business Edition, an enterprise-grade version of its acclaimed open source PBX for the Linux operating system. This version provides tested reliability of critical functions and features, tailored for small- and medium-sized business applications. Now, as to the debate about what is and is not available in an "enterprise-grade" product, I will have to defer to those who actually use Asterisk in the enterprise - I only use it for tinkering and minor voice broadcasting campaigns. -MC > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Douglas Garstang > Sent: Wednesday, March 08, 2006 7:19 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic > > I can't be bothered looking for the link right now, but it's definitely > stated somewhere on Digium's website. > > -----Original Message----- > From: Alexander Lopez [mailto:[EMAIL PROTECTED] > Sent: Tuesday, March 07, 2006 3:34 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp > traffic > > > To retort, Digium has ever to my knowledge, stamped an 'Enterprise > Grade' mark on the product. If you are worried about a single point of > failure you may want to replace your toaster. > > Asterisk is missing a 'few features' no doubt about it, but it is open > source, it will be a welcome addition if you would like to add > multi-homing support in, might as well do media multi-homing with call > diversity. This will definably be a non-trivial re-architecture of the > core. > > The 'missing a few features' way of thinking is what has made Asterisk > what it is today. > > > -----Original Message----- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Douglas Garstang > > Sent: Tuesday, March 07, 2006 11:46 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp > traffic > > > > Pardon my candour, but for a product Digium calls 'enterprise grade' > it > > sure seems to be missing a few features. > > > > -----Original Message----- > > From: Alexander Lopez [mailto:[EMAIL PROTECTED] > > Sent: Tuesday, March 07, 2006 9:39 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp > > traffic > > > > > > Asterisk does not like multiple interfaces in the way you are > configured. > > You can either: > > > > A) use the bindaddr in the sip.conf to limit where the packsge come > and > > go. > > > > B) use an outside traffic manager > > > > Look up the archives, kpf explained why this would not work, as > asterisk > > can't do load balancing at this time > > > > > > -----Original Message----- > > From: "Robert Webb" <[EMAIL PROTECTED]> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk- > > [EMAIL PROTECTED]> > > Sent: 3/7/06 11:27 AM > > Subject: Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp > traffic > > > > > > On Tue, 7 Mar 2006 09:12:25 -0700 > > "Douglas Garstang" <[EMAIL PROTECTED]> wrote: > > > I have a configuration where RTP traffic is going out > > >interface pub0, and coming back into through pub1. > > > I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an > > >shows: > > > > > > udp 0 788 0.0.0.0:5060 0.0.0.0:* > > > > > > which means that Asterisk is listening on all addresses > > >(on all interfaces?). > > > > > > Anyway, when the RTP traffic comes back in on interface > > >pub0, Asterisk does nothing with it. A 'rtp debug' shows > > >it's receiving the RTP packets, it just seems it does > > >nothing with them. > > > > > > Anyone seen this? > > > > > > Doug. > > > > > > > > > > I thought all RTP was controlled through rtp.conf and only > > the SIP traffic was controlled through SIP.conf. I am not > > sure what settings, beside the RTP port range, you can out > > into the rtp.conf though. > > > > Robert > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users