[Asterisk-Dev] Re: Asterisk-Dev Digest, Vol 12, Issue 68

Tue Jul 26 06:49:07 MST 2005

Dear Kevin,

I didn't mean it that way.  I am sorry that you miss understood.

Yes, you did point me in the right direction and I am very grateful for that.

I mean in the future, when you need an urgrent fix and want a deeper
understanding of the inner workings of asterisk. I just mean that we
should run a debug by yourself first.

Very sorry. :(

Kengie

On 7/26/05, asterisk-dev-request at lists.digium.com
<asterisk-dev-request at lists.digium.com> wrote:
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> Today's Topics:
> 
>    1. Re: DTMF Pass through (Transparently) (Steve Underwood)
>    2. Cluecon - Who's going ? (Terry Moore-Read)
>    3. Re: help needed (Michael K. Rodriguez)
>    4. Re: DTMF Pass through (Transparently) (Kevin P. Fleming)
>    5. Re: Re: Marco and Realtime Extension Problem [SOLVED]
>       (Kevin P. Fleming)
> 
> 
> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Mon, 25 Jul 2005 23:23:57 +0800
> From: Steve Underwood <steveu at coppice.org>
> Subject: Re: [Asterisk-Dev] DTMF Pass through (Transparently)
> To: RHartmann at nnamtraH.com,     Asterisk Developers Mailing List
>         <asterisk-dev at lists.digium.com>
> Message-ID: <42E5040D.1080901 at coppice.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> Ronald Hartmann wrote:
> 
> > Good Day Developers,
> >
> >
> >
> >             Thanks for all you do for asterisk.
> >
> >
> >
> >             I am desperately seeking your advise on DTMF Passthrough.
> >
> >
> >
> >             I have an application where DTMF Signalling must be
> > received exactly the same way it was transmitted.
> >
> Then your application is pretty well broken. *Many* things do not pass
> DTMF with the original timing - cellular networks, many digital
> keysystems, etc.
> 
> >              The problem is that asterisk is listening and then when
> > it hears DTMF it retransmits the DTMF.  Is it possible
> >
> >             To get the DTMF to be transparently passed through?  If
> > not are any of you able or willing to offer your expertise for hire?
> >
> If you use u-law or A-law, and don't have problems with dropped packets,
> you can pass the DTMF transparently. It won't work with lower bit rate
> codecs, though. No consulting can fix that.
> 
> Regards,
> Steve
> 
> 
> 
> 
> ------------------------------
> 
> Message: 2
> Date: Mon, 25 Jul 2005 08:29:35 -0700
> From: "Terry Moore-Read" <tmoore at lukins.com>
> Subject: [Asterisk-Dev] Cluecon - Who's going ?
> To: <asterisk-dev at lists.digium.com>,    <asterisk-users at lists.digium.com>
> Message-ID: <s2e4a2fd.031 at 10.1.1.2>
> Content-Type: text/plain; charset=US-ASCII
> 
> I'm relatively new to Asterisk and I'm hoping attending Cluecon will be
> a good way to get up to speed on the project and hear about what others
> are doing with it.
> 
> We currently use a Cisco IP phone system at my office, although I just
> added an asterisk box to provide soft phones to our travelling users.
> (IAX2 is a lot easier to get through firewalls than cisco's protocols).
> 
> Terry Moore-Read
> Lukins & Annis, P.S.
> Spokane, WA
> 
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> 
> ------------------------------
> 
> Message: 3
> Date: Mon, 25 Jul 2005 10:40:54 -0500
> From: "Michael K. Rodriguez" <michael at voipalliance.net>
> Subject: Re: [Asterisk-Dev] help needed
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <BF0A7236.10598%michael at voipalliance.net>
> Content-Type: text/plain;       charset="US-ASCII"
> 
> I had a similar problem when asterisk executed a perl AGI and exiting with
> 0. Turns out I had a syntax problem.
> 
> I would be helpful if the script was listed as well.
> 
> -Michael
> 
> 
> 
> 
> On 7/25/05 9:34 AM, "Hoai-Anh Ngo-Vi" <Hoai-Anh.Ngo-Vi at mcn-tele.com> wrote:
> 
> > Dear all
> >
> > I am using Asterisk 1.0.7 on Debian Sarge with Kernel 2.6.8-2-686.
> >
> > The Problem: I've written a C programm and placed it into
> > /usr/share/asterisk/agi-bin, where the AGI scripts should be placed.
> > Extension.conf edited as follows:
> > ...
> > exten => 336, 4, AGI(a.out)
> >
> > When dialing 336 I've got those messages from Asterisk:
> > ...
> >     -- Executing AGI("SIP/hoaianh-09f2", "a.out") in new stack
> >     -- Launched AGI Script /usr/share/asterisk/agi-bin/a.out
> >     -- AGI Script a.out completed, returning 0
> > ...
> >
> > But Asterisk didn't really launch that a.out programm (I didn't get any
> > message via CLI> console, that programm would have put some messages into CLI>
> > console via stderr if it ran correctly).
> >
> > Any idea?
> >
> > Thanks for help.
> >
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> 
> 
> ------------------------------
> 
> Message: 4
> Date: Mon, 25 Jul 2005 10:43:43 -0500
> From: "Kevin P. Fleming" <kpfleming at digium.com>
> Subject: Re: [Asterisk-Dev] DTMF Pass through (Transparently)
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <42E508AF.6070509 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> Ronald Hartmann wrote:
> 
> >             I have an application where DTMF Signalling must be received
> > exactly the same way it was transmitted.
> 
> In what environment? Zap<->Zap? Zap<->SIP? ???
> 
> >             The problem is that asterisk is listening and then when it
> > hears DTMF it retransmits the DTMF.  Is it possible
> >             To get the DTMF to be transparently passed through?  If not
> > are any of you able or willing to offer your expertise for hire?
> 
> Anything is possible, but may be completely impractical. Also, as
> already mentioned in the previous thread where this was brought up (was
> that you?), most digital PBXes don't generate predictable DTMF timing
> either.
> 
> 
> ------------------------------
> 
> Message: 5
> Date: Mon, 25 Jul 2005 10:27:47 -0500
> From: "Kevin P. Fleming" <kpfleming at digium.com>
> Subject: Re: [Asterisk-Dev] Re: Marco and Realtime Extension Problem
>         [SOLVED]
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <42E504F3.5030705 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> Kenige Ho wrote:
> 
> > I hope that this will help some people when there isn't any one to help you.
> 
> That's rather rude; I responded to your original message with enough
> information to help you arrive at this exact same point.
> 
> 
> ------------------------------
> 
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> 
> End of Asterisk-Dev Digest, Vol 12, Issue 68
> ********************************************
>