-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Thursday, January 16, 2014 8:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?
On 1/16/2014 6:57 PM, Dan Austin wrote: > Patrick Lists wrote: >> On 16-01-14 21:37, Gergely Kiss wrote: >>> Dear List, >>> >>> I'm about to build an Asterisk 11.7 based PBX from scratch for our >>> company. I'm in the middle of the planning phase and it turned out >>> that our VoIP provider prefers H.323 protocol for handling voice >>> calls (while SIP is also supported as "plan B"). >> It's SIP everywhere and anyone who requires you, in 2014, to use >> H.323 should get a clue. Avoid them or at least demand SIP > Bah. There is nothing wrong with a working H.323 stack. Just > assuming that they will have a working SIP stack because of the date > can lead to heartache. > >>> As I never worked with H.323 channels in Asterisk earlier, I'm not >>> sure if it's stable enough to be used in production. >> No idea. Maybe someone else with H.323 experience will respond. AFAIK >> it's a dead-end. > The ooh323 channel has been fairly reliable in our use case, which > involve connecting to a commercial IP PBX with crud SIP support. Only > you can tell if it will work for you however, as sadly many times new > core features only get tested against the SIP channel(s), or worse > only implemented there as well. Our current Asterisk version is > 11.5.1 > > Dan > > > Sorry, have nothing to say of 11.5 but OOH323 works great in 1.8. I use it as an Avaya IP Office trunk. No problems. As you observed for yourself when you researched the topic there is not a lot of help available, and Asterisk team prefers to make everybody think that SIP is the only viable call setup protocol around. They kind of not talking a lot about their own IAX any more. The official H.323 is abandoned. OOH323 is being supported by a very capable and responsive guy. He does not frequent the user list as he subscribes to the developer list, so I normally transfer the help inquiries to him if there is no traction here. -Vladimir Hey Vladimir, can you share a bit about the ooH323 trunk to IPO configuration that's stable? I've tried a few different setups on both sides and wound up using a PRI to do it. A call from the IPO to Asterisk (1.6.2 at the time) would crash the Asterisk box or not work at all. I'd love to be able to offer this to my IPO and CM customers! Thank you! Pat... -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users