[asterisk-dev] Re: asterisk-dev Digest, Vol 25, Issue 36
jdruin at bellsouth.net
jdruin at bellsouth.net
Thu Aug 10 20:08:50 MST 2006
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From: asterisk-dev-request at lists.digium.com
Date: Thu, 10 Aug 2006 20:01:53
To:asterisk-dev at lists.digium.com
Subject: asterisk-dev Digest, Vol 25, Issue 36
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Today's Topics:
1. Re: Re: Zaptel drivers Part 2
(Tim Johnson - Eclipse Electronic Systems)
2. Re: Re: Re: Zaptel drivers Part 2 (Steven Critchfield)
3. Channel Driver getting which mailbox was activated (Paulo Garcia)
4. Re: Channel Driver getting which mailbox was activated
(BJ Weschke)
5. Bugid 5162: Packetization (Dan Austin)
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Message: 1
Date: Thu, 10 Aug 2006 15:12:23 -0500
From: Tim Johnson - Eclipse Electronic Systems
<tjohnson at eclipse.sigint.com>
Subject: [asterisk-dev] Re: Re: Zaptel drivers Part 2
To: asterisk-dev at lists.digium.com
Message-ID: <44DB9327.1000602 at eclipse.sigint.com>
Content-Type: text/plain; charset="iso-8859-1"
I guess what I'm really trying to do is read data that has been received
on this interface in a specific channel. Yes, the data is audio, but I
don't care if the system recognizes it as audio or not since my code is
going to be dealing with sending it out to the sound card. Right now,
all I want to do is send 0x55555555 (for example) into the Digium E1
card on span 1 channel 1, and be able to read this data (or a variation
that I can work with) out with the driver.
Say all I wanted to do was send plain old binary data across this link,
into that Digium/Zaptel/Pri card, how can I access it? Is it even
possible to access it? Is there document that helps explain this, or
some code that might be doing something similar?
Thanks for all your help.
-Tim
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Message: 2
Date: Thu, 10 Aug 2006 15:18:38 -0500
From: Steven Critchfield <critch at basesys.com>
Subject: Re: [asterisk-dev] Re: Re: Zaptel drivers Part 2
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <1155241118.508.30.camel at localhost.localdomain>
Content-Type: text/plain
On Thu, 2006-08-10 at 15:12 -0500, Tim Johnson - Eclipse Electronic
Systems wrote:
> I guess what I'm really trying to do is read data that has been received
> on this interface in a specific channel. Yes, the data is audio, but I
> don't care if the system recognizes it as audio or not since my code is
> going to be dealing with sending it out to the sound card. Right now,
> all I want to do is send 0x55555555 (for example) into the Digium E1
> card on span 1 channel 1, and be able to read this data (or a variation
> that I can work with) out with the driver.
>
> Say all I wanted to do was send plain old binary data across this link,
> into that Digium/Zaptel/Pri card, how can I access it? Is it even
> possible to access it? Is there document that helps explain this, or
> some code that might be doing something similar?
Maybe you would be better suited to using clear channels. Remove any
telephony signalling from the channels. Then you might be able to go
back to using the /dev/zap/1 method for reading the data. Just remember
that there isn't a buffer so you need to be able to service the channel
quickly or you may lose audio over the link.
--
Steven Critchfield <critch at basesys.com>
------------------------------
Message: 3
Date: Thu, 10 Aug 2006 17:45:56 -0300
From: "Paulo Garcia" <paulo.astdev at gmail.com>
Subject: [asterisk-dev] Channel Driver getting which mailbox was
activated
To: asterisk-dev at lists.digium.com
Message-ID:
<b850d66b0608101345o310b16a0u2369995e05854e65 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hello,
I'm developing a channel driver for Asterisk (1.2.x) and I have the
following extension for tests:
exten => _45XXXX,1,Dial(SIP/${EXTEN:2},60,tT)
exten => _45XXXX,n,VoiceMail(u${EXTEN:2})
exten => _45XXXX,n,Hangup
When the SIP extension is busy, the VoiceMail application is activated but I
need to know within my channel driver which specific mailbox name was
called. Is it possible?
Thanks in advance!
Paulo
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Message: 4
Date: Thu, 10 Aug 2006 19:55:44 -0400
From: "BJ Weschke" <bweschke at gmail.com>
Subject: Re: [asterisk-dev] Channel Driver getting which mailbox was
activated
To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
Message-ID:
<79cf6330608101655y4b588b23o623b1e6df5a71ebd at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
On 8/10/06, Paulo Garcia <paulo.astdev at gmail.com> wrote:
> Hello,
>
> I'm developing a channel driver for Asterisk (1.2.x) and I have the
> following extension for tests:
>
> exten => _45XXXX,1,Dial(SIP/${EXTEN:2},60,tT)
> exten => _45XXXX,n,VoiceMail(u${EXTEN:2})
> exten => _45XXXX,n,Hangup
>
> When the SIP extension is busy, the VoiceMail application is activated but I
> need to know within my channel driver which specific mailbox name was
> called. Is it possible?
>
> Thanks in advance!
>
For what? MWI? No. You need to set this in the channel config like
sip.conf does and then go and try to establish whether or not you
should signal an MWI. A channel driver is at far too low a layer
within the core to understand what application it's presently serving,
let alone, what arguments were given to that application.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
------------------------------
Message: 5
Date: Thu, 10 Aug 2006 20:02:02 -0700
From: "Dan Austin" <Dan_Austin at Phoenix.com>
Subject: [asterisk-dev] Bugid 5162: Packetization
To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
Message-ID:
<B0CF4196F21DC0448367514774331AB701847DD2 at scl-exch2k3.phoenix.com>
Content-Type: text/plain; charset="us-ascii"
I've been trying to keep anthm's original patch up to date with svn, and
avoiding trying to address the modifications needed to make it IAX
friendly.
Not because I didn't want to, but because I suspected the result would
be
garbage. After exchanging some ideas with mantis user SB, I'm not so
sure.
His ideas closely matched my own, and even our first passes at the code
looked
similar.
Attached is a patch that demonstates the basic direction. This is NOT
1.4
Material, it does require trivial changes to a couple header files.
Still
I'd like to find out if this is the wrong direction while it is early.
High level details-
1. Extend AST_FORMAT_LIST to include these values-
Default payload in bytes, default payload in ms
minimum payload in ms, maximum payload in ms and
payload incriment.
2. Move the function ast_rtp_lookup_smoother from rtp.c
to frame.c and rename it to ast_lookup_smoother
3. Modify ast_lookup_smoother to use AST_FORMAT_LIST instead
of RTP_TABLE
The attached patch is a hybrid between where I think it should go and
anthm's original. In fact other than loading the values from the config
files and a change to chan_sip to confirm the values are properly
stored,
the code does not use the vaules.
If this approach is acceptable, I'll continue to work with SB to finish
the RTP implimentation and the see what it will take to make IAX happy.
So comments are more that welcome. I am pretty sure my C is lacking,
but
syntax and style can be fixed if the design is valid.
Thanks,
Dan
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