[asterisk-dev] Re: SIP to IAX (Jeremy McNamara)

Thu Oct 5 00:33:50 MST 2006

In that case, I wonder how an SIP phone which is use SIP protocol to set up
a call session can communicate with a SIP phone which is use H.323 if there
is no module to convert two types of message
Actually, we can use extension.conf to redirect from SIP channel to H.323
channel but we really need an converted module or converted function. Is
that right?

@Jeremy: Thank for your answer so much,

Regards,
Thong Lam 
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Today's Topics:

   1. Re: What's the best source for architectural	understanding?
      (Jay R. Ashworth)
   2. Re: Another bounty - app_reload (Matthew Rubenstein)
   3. Re: app_reload? (Kristian Kielhofner)
   4. Re: Re: app_reload? (Steve Edwards)
   5. Re: app_reload? (Jeremy McNamara)
   6. SIP to IAX (Thong Lam Hai)
   7. Re: SIP to IAX (Jeremy McNamara)
   8. Re: autoconf issues for FreeBSD (Luigi Rizzo)

----------------------------------------------------------------------

Message: 1
Date: Thu, 5 Oct 2006 00:03:23 -0400
From: "Jay R. Ashworth" <jra at baylink.com>
Subject: Re: [asterisk-dev] What's the best source for architectural
	understanding?
To: asterisk-dev at lists.digium.com
Message-ID: <20061005040323.GK23570 at cgi.jachomes.com>
Content-Type: text/plain; charset=us-ascii

On Wed, Oct 04, 2006 at 10:49:34PM -0500, Moises Silva wrote:
> >So, one centralized dispatcher looks for both types of off-hooks on all
> >channels?

> No, for each telephony technology must exists a channel driver that
> will "listen" or "expect" for incoming calls.

Ah.

>                                                  Once detects an incoming
> call, will try to authenticate the call against its devices configured
> in its configuration file (sip.conf, iax.conf, zapata.conf,
> unicall.conf etc), and will start a new thread in the PBX (extensions
> matching) for handling that new call. So, you have at least 1 listener
> thread for each telephony technology. And 1 thread for each call
> executing extensions in the dial plan.

Aha.

So there is actually an independent thread running for each active
call, that you could inspect and log and manhandle.  Good.  Glad to
hear that.  Thanks for the clarification.

Cheers,
-- jra
-- 
Jay R. Ashworth
jra at baylink.com
Designer                          Baylink                             RFC
2100
Ashworth & Associates        The Things I Think                        '87
e24
St Petersburg FL USA      http://baylink.pitas.com             +1 727 647
1274

	"That's women for you; you divorce them, and 10 years later,
	  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women_

------------------------------

Message: 2
Date: Thu, 05 Oct 2006 00:19:19 -0400
From: Matthew Rubenstein <email at mattruby.com>
Subject: Re: [asterisk-dev] Another bounty - app_reload
To: Kristian Kielhofner <kris at krisk.org>
Cc: Asterisk-Dev <asterisk-dev at lists.digium.com>
Message-ID: <1160021960.12032.31.camel at localhost.localdomain>
Content-Type: text/plain

On Wed, 2006-10-04 at 17:31 -0700, asterisk-dev-request at lists.digium.com
wrote:
> Date: Wed, 04 Oct 2006 19:02:08 -0400
> From: Kristian Kielhofner <kris at krisk.org>
> Subject: Re: [asterisk-dev] Another bounty - app_reload
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <45243D70.4010404 at krisk.org>
> Content-Type: text/plain; charset=iso-8859-1; format=flowed
> 
> Andrew Kohlsmith wrote:
> > On Wednesday 04 October 2006 15:47, Kristian Kielhofner wrote:
> > 
> >>      So, here is another bounty!  app_reload should be able to
> initiate a
> >>reload from the dialplan.  It should also be able to (optionally)
> >>specify a single module to reload, i.e. chan_sip.so, etc.
> > 
> > 
> > What's wrong with:
> > 
> > exten => 735623,1,System(/usr/sbin/asterisk -rx "reload")
> > exten => 735623,n,Hangup
> > 
> > ??
> > 
> > -A.
> 
> As I stated before:
> 
> "Yes, obviously System("asterisk -rx reload") and other variations
> would 
> work.  But does using the System command to launch another Asterisk 
> process to connect to the already running Asterisk instance over a
> UNIX 
> socket to issue a reload bother anyone else?  It sure bothers me... 
> Wouldn't having a dialplan app to call ast_module_reload be sooo much 
> simpler?"

	Actually, what seems missing from the diaplan apps is one that just
writes to the Manager API socket any passed string, and one to read from
the Manager API socket any sent data. The read command would be even
better if it could be connected to the API initially, and buffer data
sent from the Manager, for parsing by regexp (and flushing). These apps
would be especially useful in Perl Asterisk::AGI. That way dialplan and
Perl (etc) could always offer a consistent API even with arbitrary added
apps to the local installation.

ManagerWrite(<command>, <delimited-data-string>)
ManagerRead($[<channel-variable])
ManagerSubscribe($[channel-variable])

	Right now it looks like the only way to do it is to write a Perl
sub()
in each Asterisk::AGI Perl script that opens a socket and reads from it.
And maybe update the local install's pm. Much better if it's
standardized as part of the API that everyone's AGIs (or diaplans) can
call, depending on the installed Manager commands installed.

> --
> Kristian Kielhofner
> 
> 
-- 

(C) Matthew Rubenstein

------------------------------

Message: 3
Date: Thu, 05 Oct 2006 00:31:51 -0400
From: Kristian Kielhofner <kris at krisk.org>
Subject: [asterisk-dev] Re: app_reload?
To: Jeremy McNamara <jj at nufone.net>
Cc: asterisk-dev at lists.digium.com
Message-ID: <45248AB7.5020607 at krisk.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Jeremy McNamara wrote:
> Kristian Kielhofner wrote:
> 
>> I'm amazed this hasn't been thought of before...
>>
>> Thanks again, you made it happen!
> 
> 
> 
> I made app_reload report status to the calling party -  It now streams 
> various standard prompts to the caller informing them if the reload 
> succeeded or not.
> 
> If you hear 'thank you' that means the reload worked - There wasn't an 
> appropriate standard prompt that I could find, so I went with thank you.
> 
> If you hear the beeperr tone (with no thank you) the reload failed - 
> meaning either there is another reload already happening or the passed 
> value didn't match a module that was loaded.
> 
> 
> 
> Jeremy McNamara
> 
> 
> 
> P.S. I am developing on the v1.2 version so the SVN trunk version may be 
> totally borken.
> 

Jeremy,

	I need to send some prompts into Allison...  I'll add these.  What 
should she say?  How about:

Reload Successful
Reload Failed

	Does app_reload set any channel variables based on status?  I don't 
want this to turn into some kind of albatross for you, but I thought I'd 
ask :).

-- 
Kristian Kielhofner

------------------------------

Message: 4
Date: Wed, 4 Oct 2006 21:41:56 -0700 (PDT)
From: Steve Edwards <asterisk.org at sedwards.com>
Subject: Re: [asterisk-dev] Re: app_reload?
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <Pine.LNX.4.64.0610042140080.26031 at fs.sedwards.com>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed

1) Reload
2) Successful
3) Failed
4) Extensions
5) Sip
6) Iax
7) Zap
8) ...

On Thu, 5 Oct 2006, Kristian Kielhofner wrote:

> Jeremy McNamara wrote:
>> Kristian Kielhofner wrote:
>> 
>>> I'm amazed this hasn't been thought of before...
>>> 
>>> Thanks again, you made it happen!
>> 
>> 
>> 
>> I made app_reload report status to the calling party -  It now streams 
>> various standard prompts to the caller informing them if the reload 
>> succeeded or not.
>> 
>> If you hear 'thank you' that means the reload worked - There wasn't an 
>> appropriate standard prompt that I could find, so I went with thank you.
>> 
>> If you hear the beeperr tone (with no thank you) the reload failed - 
>> meaning either there is another reload already happening or the passed 
>> value didn't match a module that was loaded.
>> 
>> 
>> 
>> Jeremy McNamara
>> 
>> 
>> 
>> P.S. I am developing on the v1.2 version so the SVN trunk version may be 
>> totally borken.
>> 
>
> Jeremy,
>
> 	I need to send some prompts into Allison...  I'll add these.  What 
> should she say?  How about:
>
> Reload Successful
> Reload Failed
>
> 	Does app_reload set any channel variables based on status?  I don't 
> want this to turn into some kind of albatross for you, but I thought I'd
ask 
> :).
>
> -- 
> Kristian Kielhofner
> _______________________________________________
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> asterisk-dev mailing list
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Thanks in advance,
------------------------------------------------------------------------
Steve Edwards      sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                             Fax: +1-760-731-3000

------------------------------

Message: 5
Date: Thu, 05 Oct 2006 00:51:22 -0400
From: Jeremy McNamara <jj at nufone.net>
Subject: [asterisk-dev] Re: app_reload?
To: Kristian Kielhofner <kris at krisk.org>
Cc: asterisk-dev at lists.digium.com
Message-ID: <45248F4A.6070102 at nufone.net>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Kristian Kielhofner wrote:
>     Does app_reload set any channel variables based on status?  I don't 
> want this to turn into some kind of albatross for you, but I thought I'd 
> ask :).

No, but it is trivial to set them -   What were you thinkin?

Jeremy McNamara

------------------------------

Message: 6
Date: Thu, 5 Oct 2006 14:09:44 +0700
From: "Thong Lam Hai" <lhthong at tma.com.vn>
Subject: [asterisk-dev] SIP to IAX
To: <asterisk-dev at lists.digium.com>
Message-ID: <000301c6e84d$3d9aaa70$610ba8c0 at lhthong>
Content-Type: text/plain; charset="us-ascii"

Hi,

Where is the code to change from SIP message to IAX message in Asterisk?

How do 2 channels (such as channel_sip and channel_iax2) communicate with
each other?

Regards,

Thong Lam

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Message: 7
Date: Thu, 05 Oct 2006 03:11:03 -0400
From: Jeremy McNamara <jj at nufone.net>
Subject: Re: [asterisk-dev] SIP to IAX
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <4524B007.7020105 at nufone.net>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Thong Lam Hai wrote:
> Hi,
> 
>  
> 
> Where is the code to change from SIP message to IAX message in Asterisk?

There is no 'code'

> How do 2 channels (such as channel_sip and channel_iax2) communicate 
> with each other?

The Asterisk Dialplan (extensions.conf)

Jeremy McNamara

------------------------------

Message: 8
Date: Thu, 5 Oct 2006 00:15:33 -0700
From: Luigi Rizzo <rizzo at icir.org>
Subject: Re: [asterisk-dev] autoconf issues for FreeBSD
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <20061005001533.A41623 at xorpc.icir.org>
Content-Type: text/plain; charset=us-ascii

On Wed, Oct 04, 2006 at 08:37:45PM -0500, Kevin P. Fleming wrote:
> ----- Luigi Rizzo <rizzo at icir.org> wrote:
> > I am pretty sure that the code below (which was my patch at the time)
> > is NOT a proper fix - i would expect autoconf to deal with platform
> > issues
> > without having us deal with them on each and every port.
> 
> That fix probably won't work correctly any longer, since the change to use
ASTCFLAGS and ASTLDFLAGS for the build system.

as russel said, here i only patched CFLAGS and LDFLAGS which is
what the configure script uses. AST* variables were already (more
or less) set properly.

> autoconf itself does nothing to increase platform compatibility; it is a
tool that allows a package builder to increase compatibility. The benefits
from autoconf are derived entirely from the logic that we put in the
configure script, not autoconf itself.

i am sure this has been discussed at length many times.

I think the goal of autotools was to ease portability, both
cross-platform and within a single platform with different packages
installed.  Maybe for the latter, autoconf still makes sense.

But if you have to explicitly put in checks for what are platform
defaults, the first goal is defeated.

Besides, the "failures" are often subtle - it's not that configure
fails completely, it just gives you a suboptimal build because it
doesn't find 8 codecs and 4 libraries and so it doesn't build the
dependent modules.
And it does it so noisily (currently the output of asterisk's
configure is 332 lines!) that it is easy to miss the failures
while the text scrolls.

Also the choice of using "ancient greek" (pre-1977 shell) as the
language for expressing the logic in the configure script doesn't
help correctness: e.g.  without local variables, i think macros are
not reentrant and things like

	saved_CFLAGS="${CFLAGS}"
	...
	CFLAGS="${saved_CFLAGS}"

could easily be trashed.

> Once again (I'll agree with Russell here) it makes absolutely no sense for
/usr/local to be the standard place for libraries to be installed, but that
they are not in the search path for the compiler and linker. All this does
is require every package that wants to build on FreeBSD to require special
logic to add them to the search path, when the FreeBSD toolchain could
easily have these paths embedded into the toolchain by default. How
frustrating :-)

Yes, frustrating. But blaming the platform (or autoconf as i did above,
or my own ignorance as i often do privately) doesn't fix the build, so
let's work on the latter :)

	cheers
cheers: Command not found.

	/usr/local/cheers :)
	luigi

------------------------------

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