[asterisk-dev] Bounty

Thu Feb 25 06:27:03 CST 2010

I don´t think that you read the bugs. Several months ago I opened a bug
about version 1.6 running out of UDP ports. It has not been fixed. I will
jump to it as soon as I do´t have to restart the whole box every hour.
https://issues.asterisk.org/view.php?id=16774

On Thu, Feb 25, 2010 at 2:28 AM, Pavel Troller <patrol at sinus.cz> wrote:

> Hi!
>
> > Every packet that arrives goes through a logic, otherwise how woud we
> know
> > that a BYE arrived. We need to intercept that logic and interrupt a timer
> > that is running before it closes the call. That is how I see it. I wanted
> to
> > pay Digium to dio this today and they turned me down, even knowing that I
> > have the Business Edition. How does Digium expects that we in the
> Enterprise
> > take Asterisk seriously if there is no way to hire a developer to add
> what
> > is so obviously needed that every single use would benefit from it. Cisco
> > IOS, for instance, has this timer.  We are the industry, and Digium must
> do
> > what we need, and charge us for it. This seriously damages the whole idea
> of
> > open source telecommunications. They did not give me a price, just said
> "we
> > don't do that". This is unacceptable.
> > Yours
> > Federico
> >
>
> Am I right, that you need this timer ?
> timerb=32000                   ; Call setup timer. If a provisional
> response is not received
>                               ; in this amount of time, the call will
> autocongest
>                               ; Defaults to 64*timert1
>
> This is a standard feature in 1.6 (I don't know, whether it is in 1.4, but
> from your request here I think that it isn't). So, please, feel free to
> upgrade your system to the current version and enjoy the feature!
>  Now, I can imagine your complaints that you simply cannot upgrade for this
> reason and that reason etc. However, I'm quoting your "We are the industry,
> and
> Digium must do what we need, and charge us for it". It's not going this way
> in
> the Telco industry, believe me :-). I work as a voice services senior
> developer
> for major telco in Czech Republic. We are using many switches and other
> types of
> telco equipment from established companies (Siemens: EWSD, Nortel: DMS100
> and
> CS2K, Cisco, TTC Marconi etc.). Almost always, when we need to incorporate
> a new
> feature, our supplier tells us: This feature has been incorporated into the
> mainline image in version x.y. You're runnig p.q, where p<x. Please upgrade
> to
> x.y. They are even forcing us to upgrade to the new version even when we
> are
> satisfied with the old one, just by dropping support for it etc. I think
> that
> Digium is doing their things well, much better than those big companies,
> because they don't drop their support of older releases, they are at least
> fixing bugs etc. I'm afraid that "must" is not a good word to describe what
> they have or have not to do, at least I suppose that you didn't sign any
> agreement with them which explicitly states their duty to do it...
>  The last word: The Open Source advantage is, that you have the source code
> available and you can either fix these things yourself (it will not be so
> hard, because the solution is already there, it's just about backporting it
> to the older version), or hire somebody to do it for you. In the case of
> closed
> source, you would have to ask just Digium.
>
> With regards, Pavel
>
>
> > On Wed, Feb 24, 2010 at 8:47 PM, Alex Balashov <
> abalashov at evaristesys.com>wrote:
> >
> > > No.
> > >
> > > On 02/24/2010 08:39 PM, Denis Galvao wrote:
> > >
> > > > Is this possible to extract the ringing state from a 183 sip header?
> > > >
> > > > I mean, is this possible to know if the far end is ringing through a
> > > > 183 message?
> > > >
> > > > Denis.
> > > >
> > > > 2010/2/24, Alex Balashov<abalashov at evaristesys.com>:
> > > >> Who told you that you're always going to get a 180 Ringing reply?
> > > >>
> > > >> Most providers that provide PSTN trunking will give you ringback as
> > > >> in-band via an early dialog (183 Session in Progress).  With some
> > > >> calls, you may just get a provisional 100 Trying reply and then
> > > >> nothing until a sudden 200 OK.  There are many possible flows and
> > > >> scenarios.
> > > >>
> > > >> On 02/24/2010 07:59 PM, CDR wrote:
> > > >>
> > > >>> I need a new Timeout parameter added to the Dial application, for
> SIP
> > > >>> dialing. The new timeout would be "first-ring" timeout, as opposed
> to
> > > >>> timeout for connection. If we don't get a 180 Ringing message
> before a
> > > >>> certain amount of seconds, the call fails. This a needed addition
> to
> > > >>> Asterisk. I need this in version 1.4 and cannot wait the normal
> time
> > > for
> > > >>> a "new feature" process to complete. The rationale is clear: many
> > > >>> carriers will hold the call indefinitely, instead of returning a
> 503.
> > > If
> > > >>> the call is ringing, then I don't care if it rings for 60 seconds,
> but
> > > >>> if there is no ringback before 6 seconds, I need yo try another
> carrier
> > > >>> and move on.
> > > >>>
> > > >>> Please contact me at nine five four 444 seven 4 zero 8
> > > >>>
> > > >>
> > > >>
> > > >> --
> > > >> Alex Balashov - Principal
> > > >> Evariste Systems LLC
> > > >>
> > > >> Tel    : +1 678-954-0670
> > > >> Direct : +1 678-954-0671
> > > >> Web    : http://www.evaristesys.com/
> > > >>
> > > >> --
> > > >>
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> > > >
> > >
> > >
> > > --
> > > Alex Balashov - Principal
> > > Evariste Systems LLC
> > >
> > > Tel    : +1 678-954-0670
> > > Direct : +1 678-954-0671
> > > Web    : http://www.evaristesys.com/
> > >
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