Johansson Olle E
2007-Dec-22 07:51 UTC
[asterisk-users] Summary: Upgrading to Asterisk 1.4
Friends, Thanks for all the feedback. If you have additional success stories or important issues, feel free to continue the discussion. I've learned a lot from your input. As a developer, I spend too much time in the bug tracker, working with particular bugs, so I often wonder how on earth anyone can use this buggy platform for anything business-like. It really feels good to get reports on people successfully using our software and meet Asterisk users who just love the product and handle tons of calls every hour with it. And as a developer, everything is of course more simple and you live in the future, moving forward to new features, new functions all the time based on customer requirements or feature requests in the mailing list or the bug tracker... Now over to a summary of the feedback. I'm not going deeper into bugs reported, those will be handled separately. * DON'T TOUCH MY ASTERISK PBX This discussion about the 1.4 upgrade situation has given very important feedback. First, for a lot of users there's simply no reason to upgrade a PBX everytime we release a new Asterisk. Existing installations that work should not be touched unless there's a very good reason to, like a new feature that makes business sense. Just upgrading for the cause of upgrading is a feature of the non-open software industry that gets a lot of revenue from upgrades. We developers has to accept that people appreciate our work, but decide not to upgrade every installation at every release. We might have to reconsider our support policy here, where we developers abandoned 1.2 this summer. We might need another team that runs 1.2 support in the bug tracker. * MAKE UPGRADING EASIER Another issue is to make the upgrade much smoother. We can't anticipate that people upgrade from 1.0 to 1.2 to 1.4 and read all the docs for every release. They can jump from 0.8 to 1.4. Or 1.0 to the future release of 1.6. We need to assist that and haven't made a good effort in doing so. But even for upgrades from 1.2 to 1.4, we need to be more clear about changes that are required, especially for 1.2 installations that already was upgraded from 1.0 and still use the 1.0 configuration syntax. They are going to have a broken configuration in 1.4 and this is the first time that happens in Asterisk. We need to make clear that Asterisk admins need to go through the log files in 1.2 and check all deprecation warnings. These needs to be fixed before even testing 1.4. * USE ASTERISK 1.4 FOR NEW INSTALLATIONS, PLEASE My personal goal would be to get the community to start using 1.4 for all new installations. We need to produce information to help this upgrade path. It's not about upgrading systems, since we're talking about new installations. It's about upgrading the Asterisk admins and installers - human beings. The success stories reported to me personally and on the list indicates that 1.4 is indeed ready for production and it's a great product. With that, I'm now changing my focus from SIP invite states, RTP sessions and video formats to Christmas ham purchasing, baking Christmas bread (julv?rt) and decorating the Christmas tree. Of course, you understand that there's an Asterisk asterisk on top of all those trees, right? :-) After Christmas, I'm running the new Asterisk SIP Masterclass together with Daniel Mierla here in Stockholm. He's one of the core OpenSER developers and it's going to be a great class. I'm sure we will locate a set of new interesting bugs in svn trunk during that week. I'm really looking forward to that training. (Hint: We still have a few open seats... :-) ) Greetings from a dark and cold place in Sweden, without a decent amount of snow... Have a wonderful, merry and cheerful Christmas! /Olle
Andrew Joakimsen
2007-Dec-22 09:55 UTC
[asterisk-users] Summary: Upgrading to Asterisk 1.4
We've started testing Asterisk 1.4.... 1.2 has been very stable and we have processed millions of minutes with it, SIP-to-ZAP, SIP-to-SIP and SIP-to-IAX. We've been using Asterisk over 4 years now and it has really re-invented the way me and a few others think of telephones. The only inter-op issues I've ever seen are with JerkJerk's NuFone network when they were insisting on using CVS-HEAD. How things have changed. Right now one of our main gateways runs 1.0 something its internal only passes calls from IAX to ZAP and it just works -- why change it? I have not dug into any 1.4 features yet except for T38 passthrough and it seems to work ok. I need to submit a bugreport for some issues next time I come across them. {emphasis added}What are the plans for Asterisk 1.6 in regards to furthering T.38 support?{/emphasis added}
On Saturday 22 December 2007 01:51:56 am Johansson Olle E wrote:> With that, I'm now changing my focus from SIP invite states, > RTP sessions and video formats to Christmas ham purchasing, > baking Christmas bread (julv?rt) and decorating the Christmas > tree. Of course, you understand that there's an Asterisk asterisk > on top of all those trees, right? :-)Merry Christmas! And thank you. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20071222/d335fed0/attachment.pgp
Philipp von Klitzing
2007-Dec-22 16:40 UTC
[asterisk-users] Summary: Upgrading to Asterisk 1.4
Hi!> Now over to a summary of the feedback. I'm not going deeper into bugs > reported, those will be handled separately.Looks like I am a bit late, but I'll try to add my share as well to highlight some of the issues that are invovled with 1.2 to 1.4 transition: - with the advent of the "g726aal2 troubles" my preferred codec was rendered unusable, and it still is that way because this setup is too flakey, you never know if and when garbled audio will hit you. This still does not work cleanly between 1.2 and 1.4 Asterisk boxes, with me thinking that somehow on IAX this is more troublesome than on SIP. Only alaw/ulaw (too hungry) and gsm (too sparse) are left since ilbc has the potential to crash asterisk once a while (not always, not on every box). - likewise SIP INFO DTMF worked reasonable well in Asterisk 1.2, whereas my experience is that in 1.4 one should better move (back) over to RFC2833, and when doing so don't forget about the rfc2833compensate setting. - all the transitions of the type "application --> function" can be painful and error prone, especially for what concerns the replacements for DBPut and DBGet and all the levels of () and [] and {} that are now invovled. - the GROUP_COUNT and call-limit (SIP) features saw a *lot* of changes on their path from 1.0 to 1.2 to 1.4, and I hear that for 1.6 call-limit will be touched and changed yet again. So practically every new point release does this in an entirely different fashion. By the way, the README file in asterisk-1.4 is outdated and refer to upgrade instructions from 1.0 to 1.2. Having said all of the above: Asterisk is coool and grrrrreat, and everyone involved even more so - Olle included ;-) - thank you for all the effort! Cheers & happy days, Philipp von Klitzing
At 01:51 12/22/2007, Johansson Olle E wrote: >Friends, > >We might have to reconsider our support policy here, where we >developers abandoned 1.2 this summer. We might need another >team that runs 1.2 support in the bug tracker. Pretty please, with cranberry sauce on top.
> > With that, I'm now changing my focus from SIP invite states, RTP > sessions and video formats to Christmas ham purchasing, baking > Christmas bread (julv?rt) and decorating the Christmas tree. Of > course, you understand that there's an Asterisk asterisk on top of > all those trees, right? :-) > > After Christmas, I'm running the new Asterisk SIP Masterclass > together with Daniel Mierla here in Stockholm. He's one of the core > OpenSER developers and it's going to be a great class. I'm sure we > will locate a set of new interesting bugs in svn trunk during that > week. I'm really looking forward to that training. (Hint: We still > have a few open seats... :-) ) > > Greetings from a dark and cold place in Sweden, without a decent > amount of snow... > > Have a wonderful, merry and cheerful Christmas! > > /Olle >Merry Christmas to all on the list and thank you. Tony Plack
Andrew Joakimsen wrote:> > {emphasis added}What are the plans for Asterisk 1.6 in regards to > furthering T.38 support?{/emphasis added} >If you really want further T.38 support, then you should be looking at callweaver. (An Asterisk 1.2 branch). The T.38 support appears to be a lot better than the available documentation suggests.
Andrew Joakimsen
2007-Dec-27 02:02 UTC
[asterisk-users] Summary: Upgrading to Asterisk 1.4
Sure they (all what... 3 of them?) claim t38 support but it does not work. Maybe in 2012 when they release 1.0 LOL On Dec 22, 2007 6:00 PM, Thomas Kenyon <digium at sanguinarius.co.uk> wrote:> Andrew Joakimsen wrote: > > > > {emphasis added}What are the plans for Asterisk 1.6 in regards to > > furthering T.38 support?{/emphasis added} > > > > If you really want further T.38 support, then you should be looking at > callweaver. (An Asterisk 1.2 branch). > > The T.38 support appears to be a lot better than the available > documentation suggests. > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >