[asterisk-dev] Fwd: Asterisk Sip Registration Hooks

Alec Davis sivad.a at paradise.net.nz
Fri Jul 29 17:10:42 CDT 2011


>Where does it say on the Digium website that 1.6 is the 'current stable' 
version? 

Just google "site:digium.com 1.6 stable" and pick the first hit.
https://forums.digium.com/downloads

Alec


> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com 
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of 
> Kevin P. Fleming
> Sent: Saturday, 30 July 2011 9:56 a.m.
> To: asterisk-dev at lists.digium.com
> Subject: Re: [asterisk-dev] Fwd: Asterisk Sip Registration Hooks
> 
> On 07/29/2011 05:50 PM, Mike Myhre wrote:
> 
> > That is great news! That is more than I could have hoped 
> for in a hook 
> > into asterisk. How risky is it to move from 1.6 to 1.8 though? From 
> > the digium website, I see that 1.6 is listed as the current 
> stable version.
> > The downloads page shows both as being stable. There are many other 
> > features in 1.8 that I would like to take advantage of, but haven't 
> > wanted to 'open a can of worms' in 'new bugs I don't understand'.
> 
> Where does it say on the Digium website that 1.6 is the 
> 'current stable' 
> version? If it says that anywhere, we need to get it fixed, for a few
> reasons: we never use the 'stable version' terminology 
> because it's misleading, and '1.6' is not a complete version 
> number (1.6.0, 1.6.1 and
> 1.6.2 were all 1.6.x release branches).
> 
> Asterisk 1.6.0 and 1.6.1 are completely unsupported at this 
> point; 1.6.2 is in 'security fix only' mode. Asterisk 1.8 is 
> the current fully-supported release, and Asterisk 10 has just 
> entered the beta release process.
> 
> Nobody can answer how 'risky' a move from one version to 
> another is for you, because they'd have to know every way 
> that you are using it (features, configuration, etc.). Only 
> you can answer that question, by testing it.
> 
> > There is one other issue with asterisk that I have had problems 
> > working around. It is the CDR being lost when a redirect is 
> done from the AMI.
> > Looking at the bug reports, it seems to be acknowledged but 
> not fixed.
> > Is there a chance that it is fixed in 1.8 and I just 
> haven't found it 
> > documented? Are there other options that can give me the same 
> > information (like CEL) that could handle the more complex 
> situations 
> > that may occur in AMI call manipulation? A single CDR for 
> each initial 
> > call is a lot to ask when in reality it may go from being 
> answered by 
> > a queue, transferred to an operator, sent to a conference 
> bridge and 
> > then transferred to another 3 way call. I see the ultimate 
> solution to 
> > be a master-detail call record where there is one initial CDR with 
> > multiple records (queue.log or CEL?) that explain the 
> details of each 
> > subsequent leg of the call. With 1.6, if I do an AMI 
> redirect, I don't 
> > get any CDR and that is a big problem to work around.
> 
> Yes, the intent of CEL was to address the sorts of situations 
> you are referring to here, where the call was more complex 
> that just "party A calls party B".
> 
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | 
> Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us 
> out at www.digium.com & www.asterisk.org
> 
> --
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