From gui.dev at me.com  Thu Jan  1 06:40:35 2009
From: gui.dev at me.com (gui.dev at me.com)
Date: Thu, 01 Jan 2009 08:40:35 -0600
Subject: [Freeswitch-dev] Looking for AJax dev.
Message-ID: <36B95AEF-D553-4C6D-919B-117EDFB76651@me.com>

Looking for a web application developer using AJAX etc for full-time  
opportunity. Preferably California or Michigan resident but not a  
must. Experience in programming and cross platform skills a plus.

Please contact me off list at gui.dev at me.com

Thanks...
Happy Holidays!


From mikael at bjerkeland.com  Fri Jan  2 03:38:06 2009
From: mikael at bjerkeland.com (Mikael A. Bjerkeland)
Date: Fri, 02 Jan 2009 12:38:06 +0100
Subject: [Freeswitch-dev] Audio formats without transcoding
In-Reply-To: <87f2f3b90812090834x134b93eeq2036e8ce3f1c92ed@mail.gmail.com>
References: <1c3be50f0812090610u7830391cmbf3e76f96dca9b79@mail.gmail.com>
	<7355B165-9BB4-4E81-B974-42B929CB0856@jerris.com>
	<87f2f3b90812090834x134b93eeq2036e8ce3f1c92ed@mail.gmail.com>
Message-ID: <1230896286.14994.22.camel@mikael-xpsm1530>

Michael,

let's say G722 isn't found, but .wav is, will FS revert to using
the .wav file and transcode if the file extension is not specified in
playback?

I will be creating a script to convert any file using sox to every
specified format so we can avoid transcoding as much as possible.
The script will appear at
http://wiki.freeswitch.org/wiki/Mod_native_file anytime soon.


El mar, 09-12-2008 a las 08:34 -0800, Michael Collins escribi?:
> Unfortunately mod_native_file hasn't been documented on the wiki yet.
> I'm going to work on that. However, the gist of it is this: if you
> have a file in the proper format, such as g711, then you can specify
> the file name w/o the extension and FreeSWITCH will pick the right one
> for the codec being used. You'll need to make sure that your file
> names have the proper extensions, like .pcmu for mu-law. A complete
> list is available here:
> http://www.iana.org/assignments/media-types/audio/
> 
> so if you have these files:
> /tmp/hello.wav
> /tmp/hello.pcmu
> /tmp/hello.gsm
> /tmp/hello.g729
> 
> you can specify the exact file name, in which case FS will play the
> file and do whatever transcoding is needed. Or you can specify the
> filename without the extension and FS will pick the appropriate one:
> 
> <action application="playback" data="/tmp/hello"/>
> 
> Look for an update on the wiki in the next day or two. In the meantime
> please report back if you have any issues.
> 
> -MC
> 
> On Tue, Dec 9, 2008 at 7:24 AM, Michael Jerris <mike at jerris.com> wrote:
> > yes, you can use mod_native_file.
> >
> > Mike
> >
> > On Dec 9, 2008, at 9:10 AM, Juan Jose Comellas wrote:
> >
> >> I need to support calls using G.711, G.729 and G.722 and I want to
> >> avoid transcoding as much as possible. Does FreeSWITCH support playing
> >> audios in a raw format that does not require transcoding for any of
> >> these codecs? It looks like FS provides WAV audio files with the
> >> frequencies of the supported codecs and relies on transcoding all the
> >> time. Is my analysis correct? How does everybody else handle this
> >> problem?
> >
> >
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From juanjo at comellas.org  Fri Jan  2 05:41:23 2009
From: juanjo at comellas.org (Juan Jose Comellas)
Date: Fri, 2 Jan 2009 11:41:23 -0200
Subject: [Freeswitch-dev] Audio formats without transcoding
In-Reply-To: <1230896286.14994.22.camel@mikael-xpsm1530>
References: <1c3be50f0812090610u7830391cmbf3e76f96dca9b79@mail.gmail.com>
	<7355B165-9BB4-4E81-B974-42B929CB0856@jerris.com>
	<87f2f3b90812090834x134b93eeq2036e8ce3f1c92ed@mail.gmail.com>
	<1230896286.14994.22.camel@mikael-xpsm1530>
Message-ID: <1c3be50f0901020541u6bcc0e46jf53b546a8aacc06e@mail.gmail.com>

BTW, what are you planning to use to convert WAV files to G.722?


On Fri, Jan 2, 2009 at 9:38 AM, Mikael A. Bjerkeland
<mikael at bjerkeland.com>wrote:

> Michael,
>
> let's say G722 isn't found, but .wav is, will FS revert to using
> the .wav file and transcode if the file extension is not specified in
> playback?
>
> I will be creating a script to convert any file using sox to every
> specified format so we can avoid transcoding as much as possible.
> The script will appear at
> http://wiki.freeswitch.org/wiki/Mod_native_file anytime soon.
>
>
> El mar, 09-12-2008 a las 08:34 -0800, Michael Collins escribi?:
> > Unfortunately mod_native_file hasn't been documented on the wiki yet.
> > I'm going to work on that. However, the gist of it is this: if you
> > have a file in the proper format, such as g711, then you can specify
> > the file name w/o the extension and FreeSWITCH will pick the right one
> > for the codec being used. You'll need to make sure that your file
> > names have the proper extensions, like .pcmu for mu-law. A complete
> > list is available here:
> > http://www.iana.org/assignments/media-types/audio/
> >
> > so if you have these files:
> > /tmp/hello.wav
> > /tmp/hello.pcmu
> > /tmp/hello.gsm
> > /tmp/hello.g729
> >
> > you can specify the exact file name, in which case FS will play the
> > file and do whatever transcoding is needed. Or you can specify the
> > filename without the extension and FS will pick the appropriate one:
> >
> > <action application="playback" data="/tmp/hello"/>
> >
> > Look for an update on the wiki in the next day or two. In the meantime
> > please report back if you have any issues.
> >
> > -MC
> >
> > On Tue, Dec 9, 2008 at 7:24 AM, Michael Jerris <mike at jerris.com> wrote:
> > > yes, you can use mod_native_file.
> > >
> > > Mike
> > >
> > > On Dec 9, 2008, at 9:10 AM, Juan Jose Comellas wrote:
> > >
> > >> I need to support calls using G.711, G.729 and G.722 and I want to
> > >> avoid transcoding as much as possible. Does FreeSWITCH support playing
> > >> audios in a raw format that does not require transcoding for any of
> > >> these codecs? It looks like FS provides WAV audio files with the
> > >> frequencies of the supported codecs and relies on transcoding all the
> > >> time. Is my analysis correct? How does everybody else handle this
> > >> problem?
> > >
> > >
> > > _______________________________________________
> > > Freeswitch-dev mailing list
> > > Freeswitch-dev at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > > http://www.freeswitch.org
> > >
> >
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From mikael at bjerkeland.com  Fri Jan  2 06:05:02 2009
From: mikael at bjerkeland.com (Mikael A. Bjerkeland)
Date: Fri, 02 Jan 2009 15:05:02 +0100
Subject: [Freeswitch-dev] Audio formats without transcoding
In-Reply-To: <1c3be50f0901020541u6bcc0e46jf53b546a8aacc06e@mail.gmail.com>
References: <1c3be50f0812090610u7830391cmbf3e76f96dca9b79@mail.gmail.com>
	<7355B165-9BB4-4E81-B974-42B929CB0856@jerris.com>
	<87f2f3b90812090834x134b93eeq2036e8ce3f1c92ed@mail.gmail.com>
	<1230896286.14994.22.camel@mikael-xpsm1530>
	<1c3be50f0901020541u6bcc0e46jf53b546a8aacc06e@mail.gmail.com>
Message-ID: <1230905102.14994.24.camel@mikael-xpsm1530>

Haven't thought of that. I suppose as you are asking this question since
sox doesn't support converting to or from G.722?

If that's the case and you figure out how we could do this please let me
know.


El vie, 02-01-2009 a las 11:41 -0200, Juan Jose Comellas escribi?:
> BTW, what are you planning to use to convert WAV files to G.722?
> 
> 
> On Fri, Jan 2, 2009 at 9:38 AM, Mikael A. Bjerkeland
> <mikael at bjerkeland.com> wrote:
>         Michael,
>         
>         let's say G722 isn't found, but .wav is, will FS revert to
>         using
>         the .wav file and transcode if the file extension is not
>         specified in
>         playback?
>         
>         I will be creating a script to convert any file using sox to
>         every
>         specified format so we can avoid transcoding as much as
>         possible.
>         The script will appear at
>         http://wiki.freeswitch.org/wiki/Mod_native_file anytime soon.
>         
>         
>         El mar, 09-12-2008 a las 08:34 -0800, Michael Collins
>         escribi?:
>         
>         > Unfortunately mod_native_file hasn't been documented on the
>         wiki yet.
>         > I'm going to work on that. However, the gist of it is this:
>         if you
>         > have a file in the proper format, such as g711, then you can
>         specify
>         > the file name w/o the extension and FreeSWITCH will pick the
>         right one
>         > for the codec being used. You'll need to make sure that your
>         file
>         > names have the proper extensions, like .pcmu for mu-law. A
>         complete
>         > list is available here:
>         > http://www.iana.org/assignments/media-types/audio/
>         >
>         > so if you have these files:
>         > /tmp/hello.wav
>         > /tmp/hello.pcmu
>         > /tmp/hello.gsm
>         > /tmp/hello.g729
>         >
>         > you can specify the exact file name, in which case FS will
>         play the
>         > file and do whatever transcoding is needed. Or you can
>         specify the
>         > filename without the extension and FS will pick the
>         appropriate one:
>         >
>         > <action application="playback" data="/tmp/hello"/>
>         >
>         > Look for an update on the wiki in the next day or two. In
>         the meantime
>         > please report back if you have any issues.
>         >
>         > -MC
>         >
>         > On Tue, Dec 9, 2008 at 7:24 AM, Michael Jerris
>         <mike at jerris.com> wrote:
>         > > yes, you can use mod_native_file.
>         > >
>         > > Mike
>         > >
>         > > On Dec 9, 2008, at 9:10 AM, Juan Jose Comellas wrote:
>         > >
>         > >> I need to support calls using G.711, G.729 and G.722 and
>         I want to
>         > >> avoid transcoding as much as possible. Does FreeSWITCH
>         support playing
>         > >> audios in a raw format that does not require transcoding
>         for any of
>         > >> these codecs? It looks like FS provides WAV audio files
>         with the
>         > >> frequencies of the supported codecs and relies on
>         transcoding all the
>         > >> time. Is my analysis correct? How does everybody else
>         handle this
>         > >> problem?
>         > >
>         > >
>         > > _______________________________________________
>         > > Freeswitch-dev mailing list
>         > > Freeswitch-dev at lists.freeswitch.org
>         > >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>         > >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>         > > http://www.freeswitch.org
>         > >
>         >
>         > _______________________________________________
>         > Freeswitch-dev mailing list
>         > Freeswitch-dev at lists.freeswitch.org
>         > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>         > http://www.freeswitch.org
>         
>         
>         _______________________________________________
>         Freeswitch-dev mailing list
>         Freeswitch-dev at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>         http://www.freeswitch.org
>         
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From msc at freeswitch.org  Fri Jan  2 07:09:24 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 2 Jan 2009 07:09:24 -0800
Subject: [Freeswitch-dev] Audio formats without transcoding
In-Reply-To: <1230896286.14994.22.camel@mikael-xpsm1530>
References: <1c3be50f0812090610u7830391cmbf3e76f96dca9b79@mail.gmail.com>
	<7355B165-9BB4-4E81-B974-42B929CB0856@jerris.com>
	<87f2f3b90812090834x134b93eeq2036e8ce3f1c92ed@mail.gmail.com>
	<1230896286.14994.22.camel@mikael-xpsm1530>
Message-ID: <91308E34-8FD2-40EC-BAB1-307B4F283F95@freeswitch.org>

 From my experience if the native file format is not found then it  
will not play anything, but will show a file not found error.

-MC

Sent from my iPhone

On Jan 2, 2009, at 3:38 AM, "Mikael A. Bjerkeland" <mikael at bjerkeland.com 
 > wrote:

> Michael,
>
> let's say G722 isn't found, but .wav is, will FS revert to using
> the .wav file and transcode if the file extension is not specified in
> playback?
>
> I will be creating a script to convert any file using sox to every
> specified format so we can avoid transcoding as much as possible.
> The script will appear at
> http://wiki.freeswitch.org/wiki/Mod_native_file anytime soon.
>
>
> El mar, 09-12-2008 a las 08:34 -0800, Michael Collins escribi?:
>> Unfortunately mod_native_file hasn't been documented on the wiki yet.
>> I'm going to work on that. However, the gist of it is this: if you
>> have a file in the proper format, such as g711, then you can specify
>> the file name w/o the extension and FreeSWITCH will pick the right  
>> one
>> for the codec being used. You'll need to make sure that your file
>> names have the proper extensions, like .pcmu for mu-law. A complete
>> list is available here:
>> http://www.iana.org/assignments/media-types/audio/
>>
>> so if you have these files:
>> /tmp/hello.wav
>> /tmp/hello.pcmu
>> /tmp/hello.gsm
>> /tmp/hello.g729
>>
>> you can specify the exact file name, in which case FS will play the
>> file and do whatever transcoding is needed. Or you can specify the
>> filename without the extension and FS will pick the appropriate one:
>>
>> <action application="playback" data="/tmp/hello"/>
>>
>> Look for an update on the wiki in the next day or two. In the  
>> meantime
>> please report back if you have any issues.
>>
>> -MC
>>
>> On Tue, Dec 9, 2008 at 7:24 AM, Michael Jerris <mike at jerris.com>  
>> wrote:
>>> yes, you can use mod_native_file.
>>>
>>> Mike
>>>
>>> On Dec 9, 2008, at 9:10 AM, Juan Jose Comellas wrote:
>>>
>>>> I need to support calls using G.711, G.729 and G.722 and I want to
>>>> avoid transcoding as much as possible. Does FreeSWITCH support  
>>>> playing
>>>> audios in a raw format that does not require transcoding for any of
>>>> these codecs? It looks like FS provides WAV audio files with the
>>>> frequencies of the supported codecs and relies on transcoding all  
>>>> the
>>>> time. Is my analysis correct? How does everybody else handle this
>>>> problem?
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-dev mailing list
>>> Freeswitch-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> dev
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org


From mikael at bjerkeland.com  Fri Jan  2 07:38:38 2009
From: mikael at bjerkeland.com (Mikael A. Bjerkeland)
Date: Fri, 02 Jan 2009 16:38:38 +0100
Subject: [Freeswitch-dev] Audio formats without transcoding
In-Reply-To: <91308E34-8FD2-40EC-BAB1-307B4F283F95@freeswitch.org>
References: <1c3be50f0812090610u7830391cmbf3e76f96dca9b79@mail.gmail.com>
	<7355B165-9BB4-4E81-B974-42B929CB0856@jerris.com>
	<87f2f3b90812090834x134b93eeq2036e8ce3f1c92ed@mail.gmail.com>
	<1230896286.14994.22.camel@mikael-xpsm1530>
	<91308E34-8FD2-40EC-BAB1-307B4F283F95@freeswitch.org>
Message-ID: <1230910718.14994.29.camel@mikael-xpsm1530>

Hmm, what about setting a channel variable to change this behaviour? I
might be off track, but shouldn't the file be transcoded if that's the
only way to play it?


El vie, 02-01-2009 a las 07:09 -0800, Michael S Collins escribi?:
> From my experience if the native file format is not found then it  
> will not play anything, but will show a file not found error.
> 
> -MC
> 
> Sent from my iPhone
> 
> On Jan 2, 2009, at 3:38 AM, "Mikael A. Bjerkeland" <mikael at bjerkeland.com 
>  > wrote:
> 
> > Michael,
> >
> > let's say G722 isn't found, but .wav is, will FS revert to using
> > the .wav file and transcode if the file extension is not specified in
> > playback?
> >
> > I will be creating a script to convert any file using sox to every
> > specified format so we can avoid transcoding as much as possible.
> > The script will appear at
> > http://wiki.freeswitch.org/wiki/Mod_native_file anytime soon.
> >
> >
> > El mar, 09-12-2008 a las 08:34 -0800, Michael Collins escribi?:
> >> Unfortunately mod_native_file hasn't been documented on the wiki yet.
> >> I'm going to work on that. However, the gist of it is this: if you
> >> have a file in the proper format, such as g711, then you can specify
> >> the file name w/o the extension and FreeSWITCH will pick the right  
> >> one
> >> for the codec being used. You'll need to make sure that your file
> >> names have the proper extensions, like .pcmu for mu-law. A complete
> >> list is available here:
> >> http://www.iana.org/assignments/media-types/audio/
> >>
> >> so if you have these files:
> >> /tmp/hello.wav
> >> /tmp/hello.pcmu
> >> /tmp/hello.gsm
> >> /tmp/hello.g729
> >>
> >> you can specify the exact file name, in which case FS will play the
> >> file and do whatever transcoding is needed. Or you can specify the
> >> filename without the extension and FS will pick the appropriate one:
> >>
> >> <action application="playback" data="/tmp/hello"/>
> >>
> >> Look for an update on the wiki in the next day or two. In the  
> >> meantime
> >> please report back if you have any issues.
> >>
> >> -MC
> >>
> >> On Tue, Dec 9, 2008 at 7:24 AM, Michael Jerris <mike at jerris.com>  
> >> wrote:
> >>> yes, you can use mod_native_file.
> >>>
> >>> Mike
> >>>
> >>> On Dec 9, 2008, at 9:10 AM, Juan Jose Comellas wrote:
> >>>
> >>>> I need to support calls using G.711, G.729 and G.722 and I want to
> >>>> avoid transcoding as much as possible. Does FreeSWITCH support  
> >>>> playing
> >>>> audios in a raw format that does not require transcoding for any of
> >>>> these codecs? It looks like FS provides WAV audio files with the
> >>>> frequencies of the supported codecs and relies on transcoding all  
> >>>> the
> >>>> time. Is my analysis correct? How does everybody else handle this
> >>>> problem?
> >>>
> >>>
> >>> _______________________________________________
> >>> Freeswitch-dev mailing list
> >>> Freeswitch-dev at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >>> http://www.freeswitch.org
> >>>
> >>
> >> _______________________________________________
> >> Freeswitch-dev mailing list
> >> Freeswitch-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> >> dev
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From msc at freeswitch.org  Fri Jan  2 08:15:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Jan 2009 08:15:10 -0800
Subject: [Freeswitch-dev] Audio formats without transcoding
In-Reply-To: <1230910718.14994.29.camel@mikael-xpsm1530>
References: <1c3be50f0812090610u7830391cmbf3e76f96dca9b79@mail.gmail.com>
	<7355B165-9BB4-4E81-B974-42B929CB0856@jerris.com>
	<87f2f3b90812090834x134b93eeq2036e8ce3f1c92ed@mail.gmail.com>
	<1230896286.14994.22.camel@mikael-xpsm1530>
	<91308E34-8FD2-40EC-BAB1-307B4F283F95@freeswitch.org>
	<1230910718.14994.29.camel@mikael-xpsm1530>
Message-ID: <87f2f3b90901020815i2b526862t5a695fa4e6b6a521@mail.gmail.com>

That's a reasonable expectation. However, if there is more than one file
format from which the sound could be played then a decision must be made as
to which one to use. You could default to .wav and hope that there's a .wav
file present or you could create a logic tree that looks for various file
types and chooses the most appropriate under the circumstances. Most likely
you'd need more than just a channel variable - you'd need to do a small
script in your preferred language, be it Lua, Perl, Python, JavaScript...

-MC

On Fri, Jan 2, 2009 at 7:38 AM, Mikael A. Bjerkeland
<mikael at bjerkeland.com>wrote:

> Hmm, what about setting a channel variable to change this behaviour? I
> might be off track, but shouldn't the file be transcoded if that's the
> only way to play it?
>
>
> El vie, 02-01-2009 a las 07:09 -0800, Michael S Collins escribi?:
> > From my experience if the native file format is not found then it
> > will not play anything, but will show a file not found error.
> >
> > -MC
> >
> > Sent from my iPhone
> >
> > On Jan 2, 2009, at 3:38 AM, "Mikael A. Bjerkeland" <
> mikael at bjerkeland.com
> >  > wrote:
> >
> > > Michael,
> > >
> > > let's say G722 isn't found, but .wav is, will FS revert to using
> > > the .wav file and transcode if the file extension is not specified in
> > > playback?
> > >
> > > I will be creating a script to convert any file using sox to every
> > > specified format so we can avoid transcoding as much as possible.
> > > The script will appear at
> > > http://wiki.freeswitch.org/wiki/Mod_native_file anytime soon.
> > >
> > >
> > > El mar, 09-12-2008 a las 08:34 -0800, Michael Collins escribi?:
> > >> Unfortunately mod_native_file hasn't been documented on the wiki yet.
> > >> I'm going to work on that. However, the gist of it is this: if you
> > >> have a file in the proper format, such as g711, then you can specify
> > >> the file name w/o the extension and FreeSWITCH will pick the right
> > >> one
> > >> for the codec being used. You'll need to make sure that your file
> > >> names have the proper extensions, like .pcmu for mu-law. A complete
> > >> list is available here:
> > >> http://www.iana.org/assignments/media-types/audio/
> > >>
> > >> so if you have these files:
> > >> /tmp/hello.wav
> > >> /tmp/hello.pcmu
> > >> /tmp/hello.gsm
> > >> /tmp/hello.g729
> > >>
> > >> you can specify the exact file name, in which case FS will play the
> > >> file and do whatever transcoding is needed. Or you can specify the
> > >> filename without the extension and FS will pick the appropriate one:
> > >>
> > >> <action application="playback" data="/tmp/hello"/>
> > >>
> > >> Look for an update on the wiki in the next day or two. In the
> > >> meantime
> > >> please report back if you have any issues.
> > >>
> > >> -MC
> > >>
> > >> On Tue, Dec 9, 2008 at 7:24 AM, Michael Jerris <mike at jerris.com>
> > >> wrote:
> > >>> yes, you can use mod_native_file.
> > >>>
> > >>> Mike
> > >>>
> > >>> On Dec 9, 2008, at 9:10 AM, Juan Jose Comellas wrote:
> > >>>
> > >>>> I need to support calls using G.711, G.729 and G.722 and I want to
> > >>>> avoid transcoding as much as possible. Does FreeSWITCH support
> > >>>> playing
> > >>>> audios in a raw format that does not require transcoding for any of
> > >>>> these codecs? It looks like FS provides WAV audio files with the
> > >>>> frequencies of the supported codecs and relies on transcoding all
> > >>>> the
> > >>>> time. Is my analysis correct? How does everybody else handle this
> > >>>> problem?
> > >>>
> > >>>
> > >>> _______________________________________________
> > >>> Freeswitch-dev mailing list
> > >>> Freeswitch-dev at lists.freeswitch.org
> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > >>> http://www.freeswitch.org
> > >>>
> > >>
> > >> _______________________________________________
> > >> Freeswitch-dev mailing list
> > >> Freeswitch-dev at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > >> dev
> > >> http://www.freeswitch.org
> > >
> > >
> > > _______________________________________________
> > > Freeswitch-dev mailing list
> > > Freeswitch-dev at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From steveu at coppice.org  Sun Jan  4 07:21:19 2009
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 04 Jan 2009 23:21:19 +0800
Subject: [Freeswitch-dev] mod_fax
Message-ID: <4960D3EF.1030804@coppice.org>

Hi all,

I finally started to play with mod_fax today. First, a couple of little 
observations. Although there is a config file for fax, modules.conf.xml 
doesn't contain an entry for mod_fax, and dialplan/default.xml doesn't 
contain a demo like

    <extension name="test_rxfax_stream">
      <condition field="destination_number" expression="^9011$">
    <action application="answer" />
    <action application="playback" data="silence_stream://2000"/>
    <action application="rxfax" data="rxfax.tif"/>
    <action application="hangup"/>
      </condition>
    </extension>

    <extension name="test_txfax_stream">
      <condition field="destination_number" expression="^9012$">
    <action application="txfax" data="txfax.tif"/>
    <action application="hangup"/>
      </condition>
    </extension>

as it does for other modules.

For more serious things.....

If the far end of a SIP FAX transaction sends a reinvite to switch to 
T.38, FS sends a 488 back and everything fouls up. Other boxes send back 
the previous codec as the new one to use, and everything carries on 
smoothly in audio mode. I'm not a SIP expert, so I don't know the 
details of what it says on the topic, but in the real world successful 
continuance of a call requires a response other than 488. As an aside, 
the called party should be the one to initiate an attempt to use T.38, 
but in the real world the calling party often does.

If T.38 is not available (which it isn't ever right now), and the call 
starts with a low bit rate codec, we should initiate a reinvite to use 
Alaw or ulaw. If that fails we might as well abandon the call.

mod_fax currently follows the practice of my old and crude demo programs 
for *, and has apps called rxfax and txfax. This is taking a very narrow 
view of a FAX machine, and I think is too limiting. I think the 
following is how things should be:

    - One app, probably just called FAX.
    - It will be started with a flag saying if it should act as the 
calling party or the called party.
    - The app will be given optional lists of files to send, and files 
to receive.
    - The app will do its best to exchange all the files it can, 
including the use of poll mode FAXing.

The module documentation says page by page events should be added (which 
spandsp supports), and this seems a sound idea. FAXback and other 
services might be implemented through this.

Regards,
Steve



From stephane at shimaore.net  Sun Jan  4 08:46:02 2009
From: stephane at shimaore.net (=?UTF-8?Q?St=C3=A9phane_Alnet?=)
Date: Sun, 4 Jan 2009 10:46:02 -0600
Subject: [Freeswitch-dev] mod_fax
In-Reply-To: <4960D3EF.1030804@coppice.org>
References: <4960D3EF.1030804@coppice.org>
Message-ID: <b23492cd0901040846r76108ba1k61720c2afa3050ab@mail.gmail.com>

Hello,

Some notes based on my past experience with fax-relay. Sorry for the
long-ish post.

> As an aside,
> the called party should be the one to initiate an attempt to use T.38,
> but in the real world the calling party often does.

Depending on whether CNG shows up first or CED shows up first, the
switch to fax-relay might happen one way or another. Fax server
entities also tend to bypass some steps and start in fax mode (some
even forget to negotiate a voice codec).

> If T.38 is not available (which it isn't ever right now), and the call
> starts with a low bit rate codec, we should initiate a reinvite to use
> Alaw or ulaw. If that fails we might as well abandon the call.

BTW the full specs are available for free:
http://www.itu.int/rec/T-REC-T.38/en
http://www.itu.int/rec/T-REC-T.30/en

T.38 Annex D has an example for a fax-only call, but generally
speaking, in the PSTN there's no such thing as "a fax call" (or a
"modem call"). A call always starts as a voice call, and might switch
to fax mode (and back).

The basic issue Steve mentioned is that if you negotiate (at the start
of the VoIP call) a codec that is supposed to use, say, 28kb/s
(G.729), then to respect QoS over the entire call you should only
accept fax calls that will fit within that amount of bandwidth
(accounting for IP/UDPTL overhead, that might be up to 14,400b/s for
example -- don't quote me on the numbers).

In the early days, one would renegotiate the codec to G.711
("upspeed") when a fax tone was detected (assuming all fax calls use
64kb/s); if QoS denied the bandwidth upspeed (either because of
per-call bandwidth restriction configured on the gateway, Call
Admission Control, RSVP, ..), then the call would be dropped. However,
I don't think the upspeed to G.711 is strictly required, and in a
fax-relay scenario, the hop-on and hop-off gateways could also decide
to only offer the appropriate fax rates to the fax machines
(overriding the speeds offered by the actual fax machines in the T.30
stream; see spec T.30 page 53). So if the call started as a G.729
call, the gateway(s) could "clear the bits" in T.30 for anything above
the matching bandwidth.

Where it gets tricky is that some T.38 options (for example UDPTL
redundancy) might mean that the actual bandwidth available to T.30 is
much lower than the bandwidth available to the voice codec; if you
start with G.729 and then switch to T.38 with one-time redundancy, the
"assumed bandwidth" falls to 28kb/s/2=14kb/s, so you might only be
able to drive 9,600b/s fax out of that.
Another issue is that some fax models use proprietary mechanism to
switch to higher speeds. So a fax machine from brand A might do
14,400b/s with a fax machine from brand A, regardless of what the T.30
negotiated speed was. (I never looked into the gory details, so take
this as hearsay.) However in that last case there's little you can do
anyhow -- you will most probably end up oversubscribing the bandwidth
assumed for the call in any case.


Finally, in some environments, it might be OK to go over the bandwidth
assumed for a voice call in order to get a fax call through (call
completion is more important, and the network is over-engineered to
account for this). Also, fax is only half-duplex, so in large
installations, things tend to level out statistically speaking (most
traffic in fax-relay is from the sender to the receiver; however the
caller might not be the sender). Finally, fax bandwidth usage can go
over voice codec bandwidth usage in regular scenarios -- there's
nothing restricting fax-relay UDPTL traffic to 64kb/s.

HTH,
St?phane

From egghunt at gmail.com  Sun Jan  4 14:09:17 2009
From: egghunt at gmail.com (Arnaldo de Moraes Pereira)
Date: Sun, 4 Jan 2009 20:09:17 -0200
Subject: [Freeswitch-dev] Web console
Message-ID: <ff2539090901041409l59c9431fj52416f82ebc7d4e7@mail.gmail.com>

Hello,

I've written a web console using gwt:
http://lustyscripps.wordpress.com/2009/01/04/web-console-for-freeswitch

It has auto completion support and shows heartbeat event and logging
messages (event_sink). I'll continue to work on it in my spare time, it's
usable right now as it is. If anyone wants to provide artwork, I'll be glad
to accept it.

-- 
Arnaldo M Pereira
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From kristian.kielhofner at gmail.com  Sun Jan  4 15:13:02 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Sun, 4 Jan 2009 18:13:02 -0500
Subject: [Freeswitch-dev] Web console
In-Reply-To: <ff2539090901041409l59c9431fj52416f82ebc7d4e7@mail.gmail.com>
References: <ff2539090901041409l59c9431fj52416f82ebc7d4e7@mail.gmail.com>
Message-ID: <2d9149cd0901041513n3924b6c7ked46ad36a66e9c96@mail.gmail.com>

On 1/4/09, Arnaldo de Moraes Pereira <egghunt at gmail.com> wrote:
> Hello,
>
> I've written a web console using gwt:
> http://lustyscripps.wordpress.com/2009/01/04/web-console-for-freeswitch
>
> It has auto completion support and shows heartbeat event and logging
> messages (event_sink). I'll continue to work on it in my spare time, it's
> usable right now as it is. If anyone wants to provide artwork, I'll be glad
> to accept it.
>
> --
> Arnaldo M Pereira
>

  Very cool!  Nice tool for those of us that appreciate the console
interface but sometimes get stuck behind a browser with nary a
terminal emulator in sight.

  One question (not really for you, I suppose) - what about SSL
support?  I see that you are using the FreeSwitch HTTP server.  Does
it support SSL?  I suppose one could always use stunnel or something
similar but I thought I'd ask.

  Again, nice work!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From egghunt at gmail.com  Sun Jan  4 16:27:21 2009
From: egghunt at gmail.com (Arnaldo de Moraes Pereira)
Date: Sun, 4 Jan 2009 22:27:21 -0200
Subject: [Freeswitch-dev] Web console
In-Reply-To: <2d9149cd0901041513n3924b6c7ked46ad36a66e9c96@mail.gmail.com>
References: <ff2539090901041409l59c9431fj52416f82ebc7d4e7@mail.gmail.com>
	<2d9149cd0901041513n3924b6c7ked46ad36a66e9c96@mail.gmail.com>
Message-ID: <ff2539090901041627l116e91cw868e567c800867a9@mail.gmail.com>

On Sun, Jan 4, 2009 at 9:13 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> On 1/4/09, Arnaldo de Moraes Pereira <egghunt at gmail.com> wrote:
> > Hello,
> >
> > I've written a web console using gwt:
> > http://lustyscripps.wordpress.com/2009/01/04/web-console-for-freeswitch
> >
> > It has auto completion support and shows heartbeat event and logging
> > messages (event_sink). I'll continue to work on it in my spare time, it's
> > usable right now as it is. If anyone wants to provide artwork, I'll be
> glad
> > to accept it.
> >
> > --
> > Arnaldo M Pereira
> >
>
>   Very cool!  Nice tool for those of us that appreciate the console
> interface but sometimes get stuck behind a browser with nary a
> terminal emulator in sight.
>
>  One question (not really for you, I suppose) - what about SSL
> support?  I see that you are using the FreeSwitch HTTP server.  Does
> it support SSL?  I suppose one could always use stunnel or something
> similar but I thought I'd ask.
>

mod_xml_rpc currently doesn't support SSL.


>
>  Again, nice work!


Thanks!


>
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Arnaldo M Pereira
ap at arnaldopereira.com
http://www.arnaldopereira.com
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From kristian.kielhofner at gmail.com  Sun Jan  4 16:55:36 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Sun, 4 Jan 2009 19:55:36 -0500
Subject: [Freeswitch-dev] Web console
In-Reply-To: <ff2539090901041627l116e91cw868e567c800867a9@mail.gmail.com>
References: <ff2539090901041409l59c9431fj52416f82ebc7d4e7@mail.gmail.com>
	<2d9149cd0901041513n3924b6c7ked46ad36a66e9c96@mail.gmail.com>
	<ff2539090901041627l116e91cw868e567c800867a9@mail.gmail.com>
Message-ID: <2d9149cd0901041655v45904c4bodd38ca5a0d56b127@mail.gmail.com>

On 1/4/09, Arnaldo de Moraes Pereira <egghunt at gmail.com> wrote:
>
> mod_xml_rpc currently doesn't support SSL.
>

I didn't think so.  Stunnel it is, for now... ;)

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From seven at idapted.com  Mon Jan  5 03:48:19 2009
From: seven at idapted.com (seven du)
Date: Mon, 5 Jan 2009 19:48:19 +0800
Subject: [Freeswitch-dev] Audio formats without transcoding
References: <mailman.3285.1231152305.18013.freeswitch-dev@lists.freeswitch.org>
Message-ID: <C5218788-140F-47B8-ABE1-61A0D7DD9CC0@idapted.com>

> We are using the mod_native_file. It's case sensitive from my  
> experience. I tested PCMU, PCMA and G729, work fine.
> And it also works on ringback tone, just set the channel variable to  
> ringback=/sounds/somefile will work.
> Just one thing confused me. if I set a ringback tone, the G729 coded  
> file dosn't work in the current trunk 11066,
> freeswitch doens't even  try to open it. While it works on the  
> freeswitch 1.0.1 release.
>



From steveu at coppice.org  Mon Jan  5 04:08:50 2009
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 05 Jan 2009 20:08:50 +0800
Subject: [Freeswitch-dev] mod_fax
In-Reply-To: <b23492cd0901040846r76108ba1k61720c2afa3050ab@mail.gmail.com>
References: <4960D3EF.1030804@coppice.org>
	<b23492cd0901040846r76108ba1k61720c2afa3050ab@mail.gmail.com>
Message-ID: <4961F852.2010103@coppice.org>

St?phane Alnet wrote:
> Hello,
>
> Some notes based on my past experience with fax-relay. Sorry for the
> long-ish post.
>
>   
>> As an aside,
>> the called party should be the one to initiate an attempt to use T.38,
>> but in the real world the calling party often does.
>>     
>
> Depending on whether CNG shows up first or CED shows up first, the
> switch to fax-relay might happen one way or another. Fax server
> entities also tend to bypass some steps and start in fax mode (some
> even forget to negotiate a voice codec).
>   
T.38 specifically says the called end is the one which should initiate 
T.38 negotiation. The presence of absence of CED and CNG has nothing to 
do with it. If the calling end tries to initiate T.38 its a protocol 
error, though a common one. As with most SIP implementation, the 
industry standard is "as broken as our bunch of monkeys could make it".
>   
>> If T.38 is not available (which it isn't ever right now), and the call
>> starts with a low bit rate codec, we should initiate a reinvite to use
>> Alaw or ulaw. If that fails we might as well abandon the call.
>>     
>
> BTW the full specs are available for free:
> http://www.itu.int/rec/T-REC-T.38/en
> http://www.itu.int/rec/T-REC-T.30/en
>
> T.38 Annex D has an example for a fax-only call, but generally
> speaking, in the PSTN there's no such thing as "a fax call" (or a
> "modem call"). A call always starts as a voice call, and might switch
> to fax mode (and back).
>
> The basic issue Steve mentioned is that if you negotiate (at the start
> of the VoIP call) a codec that is supposed to use, say, 28kb/s
> (G.729), then to respect QoS over the entire call you should only
> accept fax calls that will fit within that amount of bandwidth
> (accounting for IP/UDPTL overhead, that might be up to 14,400b/s for
> example -- don't quote me on the numbers).
>
> In the early days, one would renegotiate the codec to G.711
> ("upspeed") when a fax tone was detected (assuming all fax calls use
> 64kb/s); if QoS denied the bandwidth upspeed (either because of
> per-call bandwidth restriction configured on the gateway, Call
> Admission Control, RSVP, ..), then the call would be dropped. However,
> I don't think the upspeed to G.711 is strictly required, and in a
> fax-relay scenario, the hop-on and hop-off gateways could also decide
> to only offer the appropriate fax rates to the fax machines
> (overriding the speeds offered by the actual fax machines in the T.30
> stream; see spec T.30 page 53). So if the call started as a G.729
> call, the gateway(s) could "clear the bits" in T.30 for anything above
> the matching bandwidth.
>
> Where it gets tricky is that some T.38 options (for example UDPTL
> redundancy) might mean that the actual bandwidth available to T.30 is
> much lower than the bandwidth available to the voice codec; if you
> start with G.729 and then switch to T.38 with one-time redundancy, the
> "assumed bandwidth" falls to 28kb/s/2=14kb/s, so you might only be
> able to drive 9,600b/s fax out of that.
> Another issue is that some fax models use proprietary mechanism to
> switch to higher speeds. So a fax machine from brand A might do
> 14,400b/s with a fax machine from brand A, regardless of what the T.30
> negotiated speed was. (I never looked into the gory details, so take
> this as hearsay.) However in that last case there's little you can do
> anyhow -- you will most probably end up oversubscribing the bandwidth
> assumed for the call in any case.
>
>
> Finally, in some environments, it might be OK to go over the bandwidth
> assumed for a voice call in order to get a fax call through (call
> completion is more important, and the network is over-engineered to
> account for this). Also, fax is only half-duplex, so in large
> installations, things tend to level out statistically speaking (most
> traffic in fax-relay is from the sender to the receiver; however the
> caller might not be the sender). Finally, fax bandwidth usage can go
> over voice codec bandwidth usage in regular scenarios -- there's
> nothing restricting fax-relay UDPTL traffic to 64kb/s.
>   
QoS is a largely unrelated issue, in that SIP servers don't cooperate 
with QoS managers. The SIP server needs to do its best to get the calls 
through, and hasn't the slightest clue what the channel's capacity might 
be. Whilst switching to a higher bit rate might overwhelm the channel, 
the SIP server has no way of telling in advance. To stick with a low bit 
rate codec is completely useless. Renegotiating to a usable codec is the 
less worst thing we can do.

Regards,
Steve



From anthony.minessale at gmail.com  Mon Jan  5 06:17:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Jan 2009 08:17:14 -0600
Subject: [Freeswitch-dev] mod_fax
In-Reply-To: <4960D3EF.1030804@coppice.org>
References: <4960D3EF.1030804@coppice.org>
Message-ID: <191c3a030901050617g2e35a9bckb30c58b41f23c215@mail.gmail.com>

mod_fax is an unfunded work in progress so calling it crude means I guess we
are not off to a very good start.
Your input is nonetheless appreciated as the small group of 3 or 4 coders
continue to try and find time to add t.30 and t.38 support to
FreeSWITCH in our spare time with little or no help.  So eventually your
concerns will probably be addressed but
Rome was not built in a day........

You do seem to have a talent for writing.  May I suggest your volunteer your
skills on our WIKI?
http://wiki.freeswitch.org/




On Sun, Jan 4, 2009 at 9:21 AM, Steve Underwood <steveu at coppice.org> wrote:

> Hi all,
>
> I finally started to play with mod_fax today. First, a couple of little
> observations. Although there is a config file for fax, modules.conf.xml
> doesn't contain an entry for mod_fax, and dialplan/default.xml doesn't
> contain a demo like
>
>    <extension name="test_rxfax_stream">
>      <condition field="destination_number" expression="^9011$">
>    <action application="answer" />
>    <action application="playback" data="silence_stream://2000"/>
>    <action application="rxfax" data="rxfax.tif"/>
>    <action application="hangup"/>
>      </condition>
>    </extension>
>
>    <extension name="test_txfax_stream">
>      <condition field="destination_number" expression="^9012$">
>    <action application="txfax" data="txfax.tif"/>
>    <action application="hangup"/>
>      </condition>
>    </extension>
>
> as it does for other modules.
>
> For more serious things.....
>
> If the far end of a SIP FAX transaction sends a reinvite to switch to
> T.38, FS sends a 488 back and everything fouls up. Other boxes send back
> the previous codec as the new one to use, and everything carries on
> smoothly in audio mode. I'm not a SIP expert, so I don't know the
> details of what it says on the topic, but in the real world successful
> continuance of a call requires a response other than 488. As an aside,
> the called party should be the one to initiate an attempt to use T.38,
> but in the real world the calling party often does.
>
> If T.38 is not available (which it isn't ever right now), and the call
> starts with a low bit rate codec, we should initiate a reinvite to use
> Alaw or ulaw. If that fails we might as well abandon the call.
>
> mod_fax currently follows the practice of my old and crude demo programs
> for *, and has apps called rxfax and txfax. This is taking a very narrow
> view of a FAX machine, and I think is too limiting. I think the
> following is how things should be:
>
>    - One app, probably just called FAX.
>    - It will be started with a flag saying if it should act as the
> calling party or the called party.
>    - The app will be given optional lists of files to send, and files
> to receive.
>    - The app will do its best to exchange all the files it can,
> including the use of poll mode FAXing.
>
> The module documentation says page by page events should be added (which
> spandsp supports), and this seems a sound idea. FAXback and other
> services might be implemented through this.
>
> Regards,
> Steve
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Jan  5 08:23:42 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Jan 2009 10:23:42 -0600
Subject: [Freeswitch-dev] mod_fax
In-Reply-To: <191c3a030901050617g2e35a9bckb30c58b41f23c215@mail.gmail.com>
References: <4960D3EF.1030804@coppice.org>
	<191c3a030901050617g2e35a9bckb30c58b41f23c215@mail.gmail.com>
Message-ID: <191c3a030901050823k767ca8d7lb3b490e8ea7a27ca@mail.gmail.com>

I think i'm a bit overworked.

Steve is one of the small group of people I was mentioning above so I am not
sure why I am reminding him of the obvious.  We thank him profusely for his
involvement.


On Mon, Jan 5, 2009 at 8:17 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> mod_fax is an unfunded work in progress so calling it crude means I guess
> we are not off to a very good start.
> Your input is nonetheless appreciated as the small group of 3 or 4 coders
> continue to try and find time to add t.30 and t.38 support to
> FreeSWITCH in our spare time with little or no help.  So eventually your
> concerns will probably be addressed but
> Rome was not built in a day........
>
> You do seem to have a talent for writing.  May I suggest your volunteer
> your skills on our WIKI?
> http://wiki.freeswitch.org/
>
>
>
>
>
> On Sun, Jan 4, 2009 at 9:21 AM, Steve Underwood <steveu at coppice.org>wrote:
>
>> Hi all,
>>
>> I finally started to play with mod_fax today. First, a couple of little
>> observations. Although there is a config file for fax, modules.conf.xml
>> doesn't contain an entry for mod_fax, and dialplan/default.xml doesn't
>> contain a demo like
>>
>>    <extension name="test_rxfax_stream">
>>      <condition field="destination_number" expression="^9011$">
>>    <action application="answer" />
>>    <action application="playback" data="silence_stream://2000"/>
>>    <action application="rxfax" data="rxfax.tif"/>
>>    <action application="hangup"/>
>>      </condition>
>>    </extension>
>>
>>    <extension name="test_txfax_stream">
>>      <condition field="destination_number" expression="^9012$">
>>    <action application="txfax" data="txfax.tif"/>
>>    <action application="hangup"/>
>>      </condition>
>>    </extension>
>>
>> as it does for other modules.
>>
>> For more serious things.....
>>
>> If the far end of a SIP FAX transaction sends a reinvite to switch to
>> T.38, FS sends a 488 back and everything fouls up. Other boxes send back
>> the previous codec as the new one to use, and everything carries on
>> smoothly in audio mode. I'm not a SIP expert, so I don't know the
>> details of what it says on the topic, but in the real world successful
>> continuance of a call requires a response other than 488. As an aside,
>> the called party should be the one to initiate an attempt to use T.38,
>> but in the real world the calling party often does.
>>
>> If T.38 is not available (which it isn't ever right now), and the call
>> starts with a low bit rate codec, we should initiate a reinvite to use
>> Alaw or ulaw. If that fails we might as well abandon the call.
>>
>> mod_fax currently follows the practice of my old and crude demo programs
>> for *, and has apps called rxfax and txfax. This is taking a very narrow
>> view of a FAX machine, and I think is too limiting. I think the
>> following is how things should be:
>>
>>    - One app, probably just called FAX.
>>    - It will be started with a flag saying if it should act as the
>> calling party or the called party.
>>    - The app will be given optional lists of files to send, and files
>> to receive.
>>    - The app will do its best to exchange all the files it can,
>> including the use of poll mode FAXing.
>>
>> The module documentation says page by page events should be added (which
>> spandsp supports), and this seems a sound idea. FAXback and other
>> services might be implemented through this.
>>
>> Regards,
>> Steve
>>
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Mon Jan  5 16:27:59 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Jan 2009 18:27:59 -0600
Subject: [Freeswitch-dev] Audio formats without transcoding
In-Reply-To: <C5218788-140F-47B8-ABE1-61A0D7DD9CC0@idapted.com>
References: <mailman.3285.1231152305.18013.freeswitch-dev@lists.freeswitch.org>
	<C5218788-140F-47B8-ABE1-61A0D7DD9CC0@idapted.com>
Message-ID: <2131EDFC-1C31-4512-BD6E-3E33EC4A7B08@freeswitch.org>

Seven,
	I have tested this with a PCMU file.. can you tell me me how you're  
setting the ringback?

/b

On Jan 5, 2009, at 5:48 AM, seven du wrote:

>> We are using the mod_native_file. It's case sensitive from my
>> experience. I tested PCMU, PCMA and G729, work fine.
>> And it also works on ringback tone, just set the channel variable to
>> ringback=/sounds/somefile will work.
>> Just one thing confused me. if I set a ringback tone, the G729 coded
>> file dosn't work in the current trunk 11066,
>> freeswitch doens't even  try to open it. While it works on the
>> freeswitch 1.0.1 release.
>>
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From seven at idapted.com  Tue Jan  6 03:46:25 2009
From: seven at idapted.com (seven du)
Date: Tue, 6 Jan 2009 19:46:25 +0800
Subject: [Freeswitch-dev] Audio formats without transcoding
In-Reply-To: <mailman.3290.1231165070.18013.freeswitch-dev@lists.freeswitch.org>
References: <mailman.3290.1231165070.18013.freeswitch-dev@lists.freeswitch.org>
Message-ID: <A1ACDDA4-73E1-4B20-87C0-8BBE8AD167D4@idapted.com>

>> sorry, mod_native_file only works on non-PASSTHROUGH codecs.  
>> previously I modified mod_g729, so FreeSWITCH takes G729 codec as  
>> transcoding codecs, mod_native_file will open sound.G729 for  
>> ringback( and playback).  However I need to make sure both legs  
>> using g729 codec.


From huyours at 163.com  Tue Jan  6 04:39:21 2009
From: huyours at 163.com (Yours)
Date: Tue, 6 Jan 2009 20:39:21 +0800 (CST)
Subject: [Freeswitch-dev] Freeswitch-dev Digest, Vol 31, Issue 3
In-Reply-To: <mailman.3290.1231165070.18013.freeswitch-dev@lists.freeswitch.org>
References: <mailman.3290.1231165070.18013.freeswitch-dev@lists.freeswitch.org>
Message-ID: <17568440.965851231245561761.JavaMail.coremail@bj163app27.163.com>

Dear friends,

If I want save user information under conf\directory\default to database, such as mysql or sqlserver, which files should I edit? I do not know which files are controlling the user information.

Thanks a lots,

London Hood
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From msc at freeswitch.org  Tue Jan  6 11:54:02 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Jan 2009 11:54:02 -0800
Subject: [Freeswitch-dev] New FreeSWITCH release
Message-ID: <87f2f3b90901061154o125d8b13v140ab3440f1980e0@mail.gmail.com>

FYI,

If you haven't already heard, we've now released v1.0.2 of FreeSWITCH!
Please digg the new release story:
http://digg.com/software/FreeSWITCH_New_Release_For_The_New_Year

The source can be downloaded here:
http://files.freeswitch.org/ in both tar.gz and tar.bz2 formats

A Windows MSI file can be downloaded here as well:
http://files.freeswitch.org/freeswitch-1.0.2.msi

Thanks for your support and keep on FreeSWITCHing!
-MC (mercutioviz)
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From stephane at shimaore.net  Tue Jan  6 21:00:17 2009
From: stephane at shimaore.net (=?UTF-8?Q?St=C3=A9phane_Alnet?=)
Date: Tue, 6 Jan 2009 23:00:17 -0600
Subject: [Freeswitch-dev] mod_fax
In-Reply-To: <191c3a030901050617g2e35a9bckb30c58b41f23c215@mail.gmail.com>
References: <4960D3EF.1030804@coppice.org>
	<191c3a030901050617g2e35a9bckb30c58b41f23c215@mail.gmail.com>
Message-ID: <b23492cd0901062100o1b0fd6aao7cb546b3cfecde70@mail.gmail.com>

Tony, Steve, & team,

> mod_fax is an unfunded work in progress so calling it crude means I guess we
> are not off to a very good start.

One idea and one question:

- It seems that doing fax detection over RTP/RFC2833 would be fairly
easy if the RFC2833_CHARS in switch_utils.c was extended to support
values like 36 (CNG) and the range 32-35 (ANS* messages). Or more
simply if the original RFC2833 event value could be accessed directly.
For example (in Javascript) if one could write:


  function onInput( session, type, data, arg ) {
    if ( type == "dtmf" ) {
      /* uses the plain RTP NTE event */
      if ( data.nte_event == 36 ) {  /* <-- e.g. "nte_event" instead
of "digit" */
        console_log( "info", "CNG received\n" );
        /* Caller is a sending fax macine, start rx_fax, etc. */
      }
  }

 ...

  /* Play voicemail prompt and attempt fax or DTMF detection */
  session.streamFile( "somefile.wav", onInput );
  /* No DTMF or fax detected, start recording the voicemail message. */


- Is there anything to be learned from T.38 and fax in CallWeaver?
app_rxfax, app_txfax, app_t38gateway... have Steve's and Tony's names
on them. Is the design there just plain wrong? Is it a licensing
issue?

On the other hand in FS's mod_fax.c I still read "the pieces are
already in place" for T.38, even though obviously the whole story
isn't ready.  Can someone elaborate on what's already there, what's
missing (UDPTL? triggering the codec changeover?), and maybe we can
attack this piece by piece? (While using the ideas Steve put forward
in his first email about merging rxfax and txfax.)

S.


From damjan at ecntelecoms.com  Tue Jan  6 22:00:40 2009
From: damjan at ecntelecoms.com (damjan at ecntelecoms.com)
Date: Wed, 7 Jan 2009 08:00:40 +0200 (SAST)
Subject: [Freeswitch-dev] mod_fax
In-Reply-To: <b23492cd0901062100o1b0fd6aao7cb546b3cfecde70@mail.gmail.com>
References: <4960D3EF.1030804@coppice.org>
	<191c3a030901050617g2e35a9bckb30c58b41f23c215@mail.gmail.com>
	<b23492cd0901062100o1b0fd6aao7cb546b3cfecde70@mail.gmail.com>
Message-ID: <62b7b61fad8306369a4278900ef7b978.squirrel@webmail.ecntelecoms.com>

> Tony, Steve, & team,
>
>> mod_fax is an unfunded work in progress so calling it crude means I
>> guess we
>> are not off to a very good start.
>
> One idea and one question:
>
> - It seems that doing fax detection over RTP/RFC2833 would be fairly
> easy if the RFC2833_CHARS in switch_utils.c was extended to support
> values like 36 (CNG) and the range 32-35 (ANS* messages). Or more
> simply if the original RFC2833 event value could be accessed directly.
> For example (in Javascript) if one could write:
>
>
>   function onInput( session, type, data, arg ) {
>     if ( type == "dtmf" ) {
>       /* uses the plain RTP NTE event */
>       if ( data.nte_event == 36 ) {  /* <-- e.g. "nte_event" instead
> of "digit" */
>         console_log( "info", "CNG received\n" );
>         /* Caller is a sending fax macine, start rx_fax, etc. */
>       }
>   }
>
>  ...
>
>   /* Play voicemail prompt and attempt fax or DTMF detection */
>   session.streamFile( "somefile.wav", onInput );
>   /* No DTMF or fax detected, start recording the voicemail message. */

In RFC2833, support for relaying modem connect tones (CNG, [/]ANS[am] and
co) is optional, and relatively few implementations support them. It also
has to be supported on both sides for it to work. Carrying these tones
over compressing codecs distorts them, often beyond recognition - that's
why in T.38 it is the answering end, which has a clear channel to the
answering fax machine, that initiates the T.38 switchover.

Also CNG and ANS are not generated or generated incorrectly by many fax
machines, you need to look for V.21H HDLC flags, or even T.30 data in the
HDLC frames, to know for sure that it's a fax.

Damjan




From seven at idapted.com  Wed Jan  7 05:44:21 2009
From: seven at idapted.com (seven du)
Date: Wed, 7 Jan 2009 21:44:21 +0800
Subject: [Freeswitch-dev]  Audio formats without transcoding
Message-ID: <B1367F7F-1828-483C-A9A1-EAB88F096214@idapted.com>

Brian,

I tested using originate  
{ignore_early_media=true,absolute_string=g729,ringback=/sounds/ 
somefile}sofia/default/1000 &bridge({ignore_early_media=true,  
ringback=/sounds/somefile,absolute_codec_string=g729}sofia/gateways/ 
xxxx/00000)

To listen the ring back, you need set ignore_early_media, the ringback  
variable maybe not needed at A-leg


To make codec g729 work, I commented one line in switch_g729_init in  
mod_g729.c:


//	codec->flags |= SWITCH_CODEC_FLAG_PASSTHROUGH;


However, I need to carefully make sure no transcoding happens,  see


http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-January/001750.html


From stephane at shimaore.net  Wed Jan  7 17:32:37 2009
From: stephane at shimaore.net (=?UTF-8?Q?St=C3=A9phane_Alnet?=)
Date: Wed, 7 Jan 2009 19:32:37 -0600
Subject: [Freeswitch-dev] mod_fax
In-Reply-To: <62b7b61fad8306369a4278900ef7b978.squirrel@webmail.ecntelecoms.com>
References: <4960D3EF.1030804@coppice.org>
	<191c3a030901050617g2e35a9bckb30c58b41f23c215@mail.gmail.com>
	<b23492cd0901062100o1b0fd6aao7cb546b3cfecde70@mail.gmail.com>
	<62b7b61fad8306369a4278900ef7b978.squirrel@webmail.ecntelecoms.com>
Message-ID: <b23492cd0901071732t21cf60d5w3583304eccff77bd@mail.gmail.com>

> In RFC2833, support for relaying modem connect tones (CNG, [/]ANS[am] and
> co) is optional, and relatively few implementations support them.
[...]
> Also CNG and ANS are not generated or generated incorrectly by many fax
> machines, you need to look for V.21H HDLC flags, or even T.30 data in the
> HDLC frames, to know for sure that it's a fax.

Alright, I guess that from an application perspective, developers
generally are not really trying to do fax detection. They are only
trying to differentiate between a "human" and a fax machine (or
modem).
So I guess "fax detection" in that relaxed sense can be summarized as:
- on an attached audio channel (AUDIO_MODE), did we detect CNG, ANS*,
or anything that would make the software switch to fax-relay mode -- I
would assume there's some kind of event that spandsp generates in that
case(?);
- on a SIP/H.323 channel supporting T.38, did a switch-over to T.38 occur.
- (any other channel mode?)

(Not that I want to drag the discussion any further if nobody cares about it.)
S.


From gmaruzz at celliax.org  Fri Jan  9 07:05:33 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 9 Jan 2009 16:05:33 +0100
Subject: [Freeswitch-dev] Crashes on Win32, sqlite problem?
Message-ID: <7b197bef0901090705t72eefea1hcd9868bde3802475@mail.gmail.com>

On Vista 32 bit, fully updated, VC Express 2008, I often (once in a
day roughly) have crashes with:

Assertion failed: inMutex, file ..\..\sqlite\src\os_win.c, line 1619

Seems a known issue:

http://jira.freeswitch.org/browse/FSCORE-180

For me, it seems unrelated to traffic or whatever...

If I can, I'll find out more, this mail just for not forgettin :-)




Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039


From earlenceferns at gmail.com  Fri Jan  9 04:17:19 2009
From: earlenceferns at gmail.com (Earlence Fernandes)
Date: Fri, 9 Jan 2009 17:47:19 +0530
Subject: [Freeswitch-dev] FreeSwitch query
Message-ID: <6853dec80901090417j48594099wcd4e8045fdf71d08@mail.gmail.com>

Hi,

I am trying to develop a VoiceChat app for the Android platform.
I came across FreeSwitch from the XMPP site where it is listed as a Jingle
compatible server.

I have a few questions.

1. Does FreeSwitch act as a VoIP server on which I can register user
accounts?.
2. If so, can I use any SIP stack(most probably in Java) to connect to this
server and establish voice calls between two custom built clients?

Cheers,
Earlence
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From mike at jerris.com  Fri Jan  9 09:23:18 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 9 Jan 2009 12:23:18 -0500
Subject: [Freeswitch-dev] FreeSwitch query
In-Reply-To: <6853dec80901090417j48594099wcd4e8045fdf71d08@mail.gmail.com>
References: <6853dec80901090417j48594099wcd4e8045fdf71d08@mail.gmail.com>
Message-ID: <060C6229-910F-4533-AC5A-98D4C2CD56A1@jerris.com>


On Jan 9, 2009, at 7:17 AM, Earlence Fernandes wrote:

> Hi,
>
> I am trying to develop a VoiceChat app for the Android platform.
> I came across FreeSwitch from the XMPP site where it is listed as a  
> Jingle compatible server.
>
> I have a few questions.
>
> 1. Does FreeSwitch act as a VoIP server on which I can register user  
> accounts?.

yes

>
> 2. If so, can I use any SIP stack(most probably in Java) to connect  
> to this server and establish voice calls between two custom built  
> clients?
>
sure

what is the xmpp connection to sip here?

> Cheers,
> Earlence

Mike



From jgarland at jasongarland.com  Fri Jan  9 13:41:41 2009
From: jgarland at jasongarland.com (Jason Garland)
Date: Fri, 9 Jan 2009 16:41:41 -0500
Subject: [Freeswitch-dev] mod_skypiax inching forward
In-Reply-To: <7b197bef0812310658u2cea5f0p7e81ea099af9de83@mail.gmail.com>
References: <7b197bef0812310658u2cea5f0p7e81ea099af9de83@mail.gmail.com>
Message-ID: <4ca506420901091341sd916347u820c05c0485d166c@mail.gmail.com>

Can you make one that calls the FS conference bridge?

On Wed, Dec 31, 2008 at 9:58 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Hi FreeSWITCHers!
>
> mod_skypiax, the Skype compatible endpoint, is slowly inching toward
> release :-)
>
> When the demo is online (will go on and off for development), you can
> test it (so helping finding bugs) by calling with Skype the Skype
> Names:
>
> skypiax20, skypiax19, skypiax18, ...., skypiax1
>
> Happy New Year !!!
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> Company : Celliax
> Website: www.celliax.org
> Address : via Pierlombardo 9, 20135 Milano
> Country/Territory : Italy
> Business Email: gmaruzz at celliax dot org
> Cell : 39-347-2665618
> Fax : 39-02-87390039
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Fri Jan  9 15:10:01 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 10 Jan 2009 00:10:01 +0100
Subject: [Freeswitch-dev] mod_skypiax inching forward
In-Reply-To: <4ca506420901091341sd916347u820c05c0485d166c@mail.gmail.com>
References: <7b197bef0812310658u2cea5f0p7e81ea099af9de83@mail.gmail.com>
	<4ca506420901091341sd916347u820c05c0485d166c@mail.gmail.com>
Message-ID: <7b197bef0901091510i4f86116bw1dba825853f8e5a8@mail.gmail.com>

you can call one of the skypiax*, then press 1 :-)

Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Fri, Jan 9, 2009 at 10:41 PM, Jason Garland
<jgarland at jasongarland.com> wrote:
> Can you make one that calls the FS conference bridge?
>
> On Wed, Dec 31, 2008 at 9:58 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Hi FreeSWITCHers!
>>
>> mod_skypiax, the Skype compatible endpoint, is slowly inching toward
>> release :-)
>>
>> When the demo is online (will go on and off for development), you can
>> test it (so helping finding bugs) by calling with Skype the Skype
>> Names:
>>
>> skypiax20, skypiax19, skypiax18, ...., skypiax1
>>
>> Happy New Year !!!
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> =========================================
>> Company : Celliax
>> Website: www.celliax.org
>> Address : via Pierlombardo 9, 20135 Milano
>> Country/Territory : Italy
>> Business Email: gmaruzz at celliax dot org
>> Cell : 39-347-2665618
>> Fax : 39-02-87390039
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From gmaruzz at celliax.org  Mon Jan 12 04:34:19 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 12 Jan 2009 13:34:19 +0100
Subject: [Freeswitch-dev] Skypiax, Skype compatible endpoint
Message-ID: <7b197bef0901120434k16f5d910j9d11012ee243e674@mail.gmail.com>

Ciao FreeSWITCHers,

mod_skypiax is now usable, for Skype calls and finding bugs :-).

Its wiki page (http://wiki.freeswitch.org/wiki/Skypiax, at the moment
it's just the README files concatenated) begins like that:

WHAT IS SKYPIAX

This software (Skypiax) uses the Skype API but is not endorsed,
certified or otherwise approved in any way by Skype.

Skypiax is an endpoint (channel driver) that use the Skype client as
an interface to the Skype network, and allows incoming and outgoing
Skype calls from/to FreeSWITCH (that can be bridged, originated,
answered, etc. as in all other endpoints, eg sofia/SIP).

Think at Skypiax as similar to OpenZAP for analog lines: for each
channel you need an interface (a Skype client). So, for eg, for two
concurrent calls, you will need two channels, two Skype clients
running on server.

If your server's Skype client(s) has got the Skype credits, Skypiax
works for SkypeOut calls too.

You can use it from the dialplan, eg with the provided modified
"default.xml" dialplan, you can call
"sip:skype/remote_skypename__OR__skypeout_phonenumber" for calling via
the Skype network from a SIP softphone to remote_skypename or to a
phone number via SkypeOut, or you can call the "2908" extension from
any phone to be bridged to the Skype Test Call).

With the provided skypiax.conf.xml all incoming Skype calls will be
routed to the "5000" extension, the IVR in default FreeSWITCH
installation.

On Linux the Skype client uses a lot of CPU. To lower its CPU
consumption, you can use the Xvfb "fake" X server and (more important)
the snd-dummy ALSA "fake" sound driver. Scripts are provided for this.
But for a low number of channels it would works with regular X servers
and ALSA drivers.

On a Linux machine with 3GB ram and a quad core intel6600, we got no
problem with 20 concurrent calls, and plenty of room for adding more
Skypiax channels (100? not tested).

On Windows, no need to do anything special, the Skype client is lighter on CPU.


Skypiax is now pre-beta, but usable for testing and finding bugs :-).


You can download Skypiax source code with subversion with the command:

svn co http://svn.freeswitch.org/svn/freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax
mod_skypiax

then, follow the README file in the mod_skypiax directory.


More info on skypiax:

http://wiki.freeswitch.org/wiki/Skypiax

http://www.celliax.org


Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039


From rehan at supertec.com  Mon Jan 12 16:49:26 2009
From: rehan at supertec.com (Rehan Allah Wala)
Date: Mon, 12 Jan 2009 17:49:26 -0700
Subject: [Freeswitch-dev] [Freeswitch-users] Skypiax,
	Skype compatible endpoint
In-Reply-To: <7b197bef0901120434k16f5d910j9d11012ee243e674@mail.gmail.com>
References: <7b197bef0901120434k16f5d910j9d11012ee243e674@mail.gmail.com>
Message-ID: <496B82A6.3574.4468654E@rehan.supertec.com>

i am looking for a consulant to send me a quote to run this for me on amazon ec2

Rehan



> Ciao FreeSWITCHers,
> 
> mod_skypiax is now usable, for Skype calls and finding bugs :-).
> 
> Its wiki page (http://wiki.freeswitch.org/wiki/Skypiax, at the moment
> it's just the README files concatenated) begins like that:
> 
> WHAT IS SKYPIAX
> 
> This software (Skypiax) uses the Skype API but is not endorsed,
> certified or otherwise approved in any way by Skype.
> 
> Skypiax is an endpoint (channel driver) that use the Skype client as
> an interface to the Skype network, and allows incoming and outgoing
> Skype calls from/to FreeSWITCH (that can be bridged, originated,
> answered, etc. as in all other endpoints, eg sofia/SIP).
> 
> Think at Skypiax as similar to OpenZAP for analog lines: for each
> channel you need an interface (a Skype client). So, for eg, for two
> concurrent calls, you will need two channels, two Skype clients
> running on server.
> 
> If your server's Skype client(s) has got the Skype credits, Skypiax
> works for SkypeOut calls too.
> 
> You can use it from the dialplan, eg with the provided modified
> "default.xml" dialplan, you can call
> "sip:skype/remote_skypename__OR__skypeout_phonenumber" for calling via
> the Skype network from a SIP softphone to remote_skypename or to a
> phone number via SkypeOut, or you can call the "2908" extension from
> any phone to be bridged to the Skype Test Call).
> 
> With the provided skypiax.conf.xml all incoming Skype calls will be
> routed to the "5000" extension, the IVR in default FreeSWITCH
> installation.
> 
> On Linux the Skype client uses a lot of CPU. To lower its CPU
> consumption, you can use the Xvfb "fake" X server and (more important)
> the snd-dummy ALSA "fake" sound driver. Scripts are provided for this.
> But for a low number of channels it would works with regular X servers
> and ALSA drivers.
> 
> On a Linux machine with 3GB ram and a quad core intel6600, we got no
> problem with 20 concurrent calls, and plenty of room for adding more
> Skypiax channels (100? not tested).
> 
> On Windows, no need to do anything special, the Skype client is lighter on CPU.
> 
> 
> Skypiax is now pre-beta, but usable for testing and finding bugs :-).
> 
> 
> You can download Skypiax source code with subversion with the command:
> 
> svn co http://svn.freeswitch.org/svn/freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax
> mod_skypiax
> 
> then, follow the README file in the mod_skypiax directory.
> 
> 
> More info on skypiax:
> 
> http://wiki.freeswitch.org/wiki/Skypiax
> 
> http://www.celliax.org
> 
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> =========================================
> Company : Celliax
> Website: www.celliax.org
> Address : via Pierlombardo 9, 20135 Milano
> Country/Territory : Italy
> Business Email: gmaruzz at celliax dot org
> Cell : 39-347-2665618
> Fax : 39-02-87390039
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



Rehan Ahmed AllahWala
Msn/Yahoo/GoogleTalk/Email: Rehan at Rehan.com
http://www.supertec.com/ - Internet Telephony Solutions
Http://www.DIDX.net - DID Number Market Place.
Don't Remember Me ? Visit http://www.Rehan.com

~~~~~~~~~~~~~~~~~~~
"First they ignore you, then they laugh at you, then they fight you, then you win." 
By Gandhi.

"Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi



From mablendafx at yahoo.com  Mon Jan 12 06:39:07 2009
From: mablendafx at yahoo.com (Antonio Murrell)
Date: Mon, 12 Jan 2009 06:39:07 -0800 (PST)
Subject: [Freeswitch-dev] Skypiax, Skype compatible endpoint
In-Reply-To: <7b197bef0901120434k16f5d910j9d11012ee243e674@mail.gmail.com>
Message-ID: <250899.34394.qm@web32801.mail.mud.yahoo.com>

Hello,
Would it work with Asterisk? what version?

thank you




--- On Mon, 1/12/09, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:

> From: Giovanni Maruzzelli <gmaruzz at celliax.org>
> Subject: [Freeswitch-dev] Skypiax, Skype compatible endpoint
> To: freeswitch-users at lists.freeswitch.org, freeswitch-dev at lists.freeswitch.org
> Date: Monday, January 12, 2009, 6:34 AM
> Ciao FreeSWITCHers,
> 
> mod_skypiax is now usable, for Skype calls and finding bugs
> :-).
> 
> Its wiki page (http://wiki.freeswitch.org/wiki/Skypiax, at
> the moment
> it's just the README files concatenated) begins like
> that:
> 
> WHAT IS SKYPIAX
> 
> This software (Skypiax) uses the Skype API but is not
> endorsed,
> certified or otherwise approved in any way by Skype.
> 
> Skypiax is an endpoint (channel driver) that use the Skype
> client as
> an interface to the Skype network, and allows incoming and
> outgoing
> Skype calls from/to FreeSWITCH (that can be bridged,
> originated,
> answered, etc. as in all other endpoints, eg sofia/SIP).
> 
> Think at Skypiax as similar to OpenZAP for analog lines:
> for each
> channel you need an interface (a Skype client). So, for eg,
> for two
> concurrent calls, you will need two channels, two Skype
> clients
> running on server.
> 
> If your server's Skype client(s) has got the Skype
> credits, Skypiax
> works for SkypeOut calls too.
> 
> You can use it from the dialplan, eg with the provided
> modified
> "default.xml" dialplan, you can call
> "sip:skype/remote_skypename__OR__skypeout_phonenumber"
> for calling via
> the Skype network from a SIP softphone to remote_skypename
> or to a
> phone number via SkypeOut, or you can call the
> "2908" extension from
> any phone to be bridged to the Skype Test Call).
> 
> With the provided skypiax.conf.xml all incoming Skype calls
> will be
> routed to the "5000" extension, the IVR in
> default FreeSWITCH
> installation.
> 
> On Linux the Skype client uses a lot of CPU. To lower its
> CPU
> consumption, you can use the Xvfb "fake" X server
> and (more important)
> the snd-dummy ALSA "fake" sound driver. Scripts
> are provided for this.
> But for a low number of channels it would works with
> regular X servers
> and ALSA drivers.
> 
> On a Linux machine with 3GB ram and a quad core intel6600,
> we got no
> problem with 20 concurrent calls, and plenty of room for
> adding more
> Skypiax channels (100? not tested).
> 
> On Windows, no need to do anything special, the Skype
> client is lighter on CPU.
> 
> 
> Skypiax is now pre-beta, but usable for testing and finding
> bugs :-).
> 
> 
> You can download Skypiax source code with subversion with
> the command:
> 
> svn co
> http://svn.freeswitch.org/svn/freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax
> mod_skypiax
> 
> then, follow the README file in the mod_skypiax directory.
> 
> 
> More info on skypiax:
> 
> http://wiki.freeswitch.org/wiki/Skypiax
> 
> http://www.celliax.org
> 
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> =========================================
> Company : Celliax
> Website: www.celliax.org
> Address : via Pierlombardo 9, 20135 Milano
> Country/Territory : Italy
> Business Email: gmaruzz at celliax dot org
> Cell : 39-347-2665618
> Fax : 39-02-87390039
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org


      


From gmaruzz at celliax.org  Mon Jan 12 07:22:27 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 12 Jan 2009 16:22:27 +0100
Subject: [Freeswitch-dev] Skypiax, Skype compatible endpoint
In-Reply-To: <250899.34394.qm@web32801.mail.mud.yahoo.com>
References: <7b197bef0901120434k16f5d910j9d11012ee243e674@mail.gmail.com>
	<250899.34394.qm@web32801.mail.mud.yahoo.com>
Message-ID: <7b197bef0901120722r730fad60l5035895d9eb35116@mail.gmail.com>

On Mon, Jan 12, 2009 at 3:39 PM, Antonio Murrell <mablendafx at yahoo.com> wrote:
> Would it work with Asterisk? what version?

Ciao Antonio,
actually it was born as a prototype channel driver for Asterisk, but
just monochannel (one only concurrent call).

If there is interest, all the developments made for FS can be
backported to Asterisk in future, and then Skypiax will go in parallel
on the two architectures, with one source file for FS interfacing, one
for * interfacing, and the Skype API interfacing in a common file.


Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039


From sanju at 11hit.com  Mon Jan 12 21:53:50 2009
From: sanju at 11hit.com (Sanju)
Date: Mon, 12 Jan 2009 21:53:50 -0800
Subject: [Freeswitch-dev] Reg: Configuring the Freeswitch
Message-ID: <20090112215350.A9783AA6@resin17.mta.everyone.net>

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From intralanman at freeswitch.org  Tue Jan 13 07:43:41 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Tue, 13 Jan 2009 15:43:41 +0000
Subject: [Freeswitch-dev] Reg: Configuring the Freeswitch
In-Reply-To: <20090112215350.A9783AA6@resin17.mta.everyone.net>
References: <20090112215350.A9783AA6@resin17.mta.everyone.net>
Message-ID: <496CB6AD.4080801@freeswitch.org>

Questions like this should really go to the freeswitch-users list... the 
-dev list should be reserved for talk of future work or actual 
development of freeswitch or modules (or even third-party addons) but 
basic use cases should really go to the -users list. 
freeswitch-users at lists.freeswitch.org

-Ray

Sanju wrote:
> Hi,
>     Good day to you all, i am new to VOIP and freeswitch, i have 
> installed freeswitch in Redhat5.1 machine and configured it properly, 
> i had xlite in Windows and configured a SIP account to Freeswitch 
> machine , but when i make a call it doesn't proceed, when i see the 
> Wikipbx page it looks for lot of packages installation does all this 
> required is postgresql and mysql has to be compulsory to be installed 
> in freeswitch server machine,
>
> can i install freeswitch in Fedora core 10 ?
>
> please let me know with what minimal package configuration i can 
> install free switch and configure x-lite for testing the applications
>
> Thanks in advance
>
> Regards,
> Sanju.
>
>
>
>
>  
> ------------------------------------------------------------------------
> Gift Certificates
> http://www.online-gift-certificate.com
>
> This email was sent using 11hit.com free web-based email!
> http://www.11hit.com
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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> http://www.freeswitch.org
>   

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From seven at idapted.com  Thu Jan 15 02:22:46 2009
From: seven at idapted.com (seven du)
Date: Thu, 15 Jan 2009 18:22:46 +0800
Subject: [Freeswitch-dev]  mod_skypiax inching forward
Message-ID: <9B415DB5-EFEC-4393-BD5F-8681252B2178@idapted.com>

Hi,

I tested skypiax on Linux db1.veecue.com 2.6.24-19-xen (Ubuntu hardy  
in Xen), after I load the mod_skypiax, the console will stuck.

I'm not sure, anything is following the wiki. However, On the Desktop  
computer configuring skype, there are multiple snd drivers,

default
HW dummy
other

I chosed the second,  so the last few lines in  config.xml in / 
root/.Skype/idapted_voip_1/  like

   <UI>
     <API>
       <Authorizations>skypiax</Authorizations>
       <BlockedPrograms></BlockedPrograms>
     </API>
     <CaptureDevice>2</CaptureDevice>
     <RingDevice>2</RingDevice>
     <SoundDevice>2</SoundDevice>
   </UI>



# sh startskype.sh
ERROR: Module snd_hda_intel does not exist in /proc/modules
error opening security policy file /etc/X11/xserver/SecurityPolicy
sh: /usr/bin/xkbcomp: not found
Could not init font path element /usr/share/fonts/X11/cyrillic,  
removing from list!
Could not init font path element /usr/share/fonts/ 
X11/100dpi/:unscaled, removing from list!
Could not init font path element /usr/share/fonts/X11/75dpi/:unscaled,  
removing from list!
Could not init font path element /usr/share/fonts/X11/Type1, removing  
from list!
Could not init font path element /usr/share/fonts/X11/100dpi, removing  
from list!
Could not init font path element /usr/share/fonts/X11/75dpi, removing  
from list!
Could not init font path element /var/lib/defoma/x-ttcidfont-conf.d/ 
dirs/TrueType, removing from list!
[config/hal] couldn't initialise context: (null) ((null))

#ps aux|grep skype

root     18774  0.3  4.0  75788 32216 pts/9    Sl   17:54   0:03 /usr/ 
bin/skype --pipelogin
root     20613  0.0  0.0   1692   500 pts/9    R+   18:16   0:00 grep  
skype




2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:630 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  630  ][none      ][-1,-1,-1]  
globals.debug=0
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:632 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  632  ][none      ][-1,-1,-1]  
globals.debug=8
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:643 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  643  ][none      ][-1,-1,-1] codec- 
master globals.debug=8
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:646 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  646  ][none      ][-1,-1,-1]  
globals.dialplan=XML
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:652 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  652  ][none      ][-1,-1,-1]  
globals.context=default
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:649 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  649  ][none      ][-1,-1,-1]  
globals.destination=5000
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:655 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  655  ][none      ][-1,-1,-1]  
globals.codec_string=gsm,ulaw
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:662 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  662  ][none      ][-1,-1,-1]  
globals.codec_rates_string=8000,16000
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:635 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  635  ][none      ][-1,-1,-1]  
globals.hold_music=local_stream://moh
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:757 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  757  ][none      ][-1,-1,-1]  
interface_id=1
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:780 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  780  ][none      ][-1,-1,-1]  
name=idapted_voip_1
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:786 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  786  ][none      ][-1,-1,-1]  
Initialized XInitThreads!
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:798 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  798  ][none      ][-1,-1,-1]  
CONFIGURING interface_id=1
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:827 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  827  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:831 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  831  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].skype_user=idapted_voip_11
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:835 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  835  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:839 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  839  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:842 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  842  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].name=idapted_voip_1
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:845 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  845  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:849 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  849  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:853 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  853  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].destination=5000
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:856 load_config() rev  
11066M[(nil)|37     ][DEBUG_SKYPE  856  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default
2009-01-15 17:56:41 [NOTICE] mod_skypiax.c:857 load_config() rev  
11066M[(nil)|37     ][NOTICA  857  ][none      ][-1,-1,-1] STARTING  
interface_id=1
2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:925  
skypiax_skypeapi_thread_func() rev 11066M[(nil)|37     ][DEBUG_PBX   
925  ][idapted_voip_1][-1, 0, 0] ENTERING FUNC
2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:938  
skypiax_skypeapi_thread_func() rev 11066M[(nil)|37     ][DEBUG_SKYPE   
938  ][idapted_voip_1][-1, 0, 0] X Display ':101' opened
2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:890  
skypiax_skype_present() rev 11066M[(nil)|37     ][DEBUG_SKYPE  890  ] 
[none      ][-1,-1,-1] Skype instance found with id #2097250
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:570  
skypiax_signaling_thread_func() rev 11066M[(nil)|37     ][DEBUG_PBX   
570  ][idapted_voip_1][-1, 0, 0] ENTERING FUNC
2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:573  
skypiax_signaling_thread_func() rev 11066M[(nil)|37     ][DEBUG_SKYPE   
573  ][idapted_voip_1][-1, 0, 0] In skypiax_signaling_thread_func:  
started, p=0xa792a420
2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||OK|||
2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||PROTOCOL 6|||
2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||CONNSTATUS ONLINE|||
2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||CURRENTUSERHANDLE  
idapted_voip_1|2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||CURRENTUSERHANDLE  
idapted_voip_1|||
2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||USERSTATUS ONLINE|||
2009-01-15 17:56:42 [DEBUG] skypiax_protocol.c:471  
skypiax_skypeaudio_init() rev 11066M[(nil)|37     ][DEBUG_PBX  471  ] 
[idapted_voip_1][-1, 0, 0] EXITING FUNC





From krice at suspicious.org  Thu Jan 15 05:44:32 2009
From: krice at suspicious.org (Ken Rice)
Date: Thu, 15 Jan 2009 07:44:32 -0600
Subject: [Freeswitch-dev] Announcing the FreeSWITCH Technology Preview
	VMWare Appliance.
Message-ID: <C59499E0.4BF72%krice@suspicious.org>


FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible
with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac.

Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was
updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed
and operational. See /etc/motd on the running image for all the good
information.

We'll be unvailing a wiki page for this shortly.

For now you can get the head start by downloading this at
http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip

Have fun guys!

Ken  
krice at freeswitch.org
krice at rmktek.com





From ludovic.fouquet at bewan.com  Tue Jan 13 23:23:48 2009
From: ludovic.fouquet at bewan.com (ludovic)
Date: Wed, 14 Jan 2009 08:23:48 +0100
Subject: [Freeswitch-dev] bridge application : how to retrieve data field
Message-ID: <496D9304.8030804@bewan.com>

Hi,

I develop my own endpoint.
I have a question regarding the xml dialplan rule like  ( I take openzap 
for the example) :

<action application="bridge" data="openzap/1/1"/>

I understand that the first field of data is the mod name. The remaining 
"1/1" is useful for openzap to select the right channel,
Where is 1/1 passed in Freeswitch ? In a callback routine of the mod ? 
or elsewhere ?

Regards,

Ludovic



From d at d-man.org  Wed Jan 14 22:12:50 2009
From: d at d-man.org (Darren Schreiber)
Date: Wed, 14 Jan 2009 22:12:50 -0800
Subject: [Freeswitch-dev] Using ODBC core from FreeSWITCH module
Message-ID: <03AFD2B5818449BF8C780A9719BCD6D8@test>

Hey there,
    I'm using the ODBC abilities of FreeSWITCH for my billing module and I
am having an issue.
 
When calling:
 
                if (!(switch_odbc_handle_callback_exec(globals.master_odbc,
SQL_UPDATE_STATEMENT, nibblebill_callback, &pdata) == SWITCH_ODBC_SUCCESS)){

sometimes the call is actually failing but I am always getting an ODBC
success. I am thinking "failing" is really 0 rows updated. How should I be
checking for this occurrence?
 
- Darren
 
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From krice at freeswitch.org  Wed Jan 14 20:05:21 2009
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 14 Jan 2009 22:05:21 -0600
Subject: [Freeswitch-dev] Announcing the FreeSWITCH Technology Preview
	VMWare Appliance.
Message-ID: <C5941221.4BF27%krice@freeswitch.org>

Hey guys, 

I'm not trying to start 1 a day releases, Things just happened to fall that
way...

FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible
with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac.

Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was
updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed
and operational. See /etc/motd on the running image for all the good
information.

We'll be unvailing a wiki page for this shortly.

For now you can get the head start by downloading this at
http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip

Have fun guys!

Ken  




From gmaruzz at celliax.org  Thu Jan 15 08:03:18 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 15 Jan 2009 17:03:18 +0100
Subject: [Freeswitch-dev] mod_skypiax inching forward
In-Reply-To: <9B415DB5-EFEC-4393-BD5F-8681252B2178@idapted.com>
References: <9B415DB5-EFEC-4393-BD5F-8681252B2178@idapted.com>
Message-ID: <7b197bef0901150803p57290e6al3ac6e9422cd625c8@mail.gmail.com>

On Thu, Jan 15, 2009 at 11:22 AM, seven du <seven at idapted.com> wrote:
> Hi,
>
> I tested skypiax on Linux db1.veecue.com 2.6.24-19-xen (Ubuntu hardy
> in Xen), after I load the mod_skypiax, the console will stuck.
>

Hi Seven,
Seems that there are no errors, and I don't think is a sound problem,
at the moment.

Also, your Skype config.xml seems correct.

I see you have an old version, could you please svn update the skypiax
code, recompile, and try again?

If there still problems, can you post the debug after "console
loglevel 9", the skypiax.conf.xml, and the lsmod|grep snd?

 Ciao for now,

Giovanni


> I'm not sure, anything is following the wiki. However, On the Desktop
> computer configuring skype, there are multiple snd drivers,
>
> default
> HW dummy
> other
>
> I chosed the second,  so the last few lines in  config.xml in /
> root/.Skype/idapted_voip_1/  like
>
>   <UI>
>     <API>
>       <Authorizations>skypiax</Authorizations>
>       <BlockedPrograms></BlockedPrograms>
>     </API>
>     <CaptureDevice>2</CaptureDevice>
>     <RingDevice>2</RingDevice>
>     <SoundDevice>2</SoundDevice>
>   </UI>
>
>
>
> # sh startskype.sh
> ERROR: Module snd_hda_intel does not exist in /proc/modules
> error opening security policy file /etc/X11/xserver/SecurityPolicy
> sh: /usr/bin/xkbcomp: not found
> Could not init font path element /usr/share/fonts/X11/cyrillic,
> removing from list!
> Could not init font path element /usr/share/fonts/
> X11/100dpi/:unscaled, removing from list!
> Could not init font path element /usr/share/fonts/X11/75dpi/:unscaled,
> removing from list!
> Could not init font path element /usr/share/fonts/X11/Type1, removing
> from list!
> Could not init font path element /usr/share/fonts/X11/100dpi, removing
> from list!
> Could not init font path element /usr/share/fonts/X11/75dpi, removing
> from list!
> Could not init font path element /var/lib/defoma/x-ttcidfont-conf.d/
> dirs/TrueType, removing from list!
> [config/hal] couldn't initialise context: (null) ((null))
>
> #ps aux|grep skype
>
> root     18774  0.3  4.0  75788 32216 pts/9    Sl   17:54   0:03 /usr/
> bin/skype --pipelogin
> root     20613  0.0  0.0   1692   500 pts/9    R+   18:16   0:00 grep
> skype
>
>
>
>
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:630 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  630  ][none      ][-1,-1,-1]
> globals.debug=0
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:632 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  632  ][none      ][-1,-1,-1]
> globals.debug=8
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:643 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  643  ][none      ][-1,-1,-1] codec-
> master globals.debug=8
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:646 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  646  ][none      ][-1,-1,-1]
> globals.dialplan=XML
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:652 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  652  ][none      ][-1,-1,-1]
> globals.context=default
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:649 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  649  ][none      ][-1,-1,-1]
> globals.destination=5000
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:655 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  655  ][none      ][-1,-1,-1]
> globals.codec_string=gsm,ulaw
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:662 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  662  ][none      ][-1,-1,-1]
> globals.codec_rates_string=8000,16000
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:635 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  635  ][none      ][-1,-1,-1]
> globals.hold_music=local_stream://moh
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:757 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  757  ][none      ][-1,-1,-1]
> interface_id=1
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:780 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  780  ][none      ][-1,-1,-1]
> name=idapted_voip_1
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:786 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  786  ][none      ][-1,-1,-1]
> Initialized XInitThreads!
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:798 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  798  ][none      ][-1,-1,-1]
> CONFIGURING interface_id=1
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:827 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  827  ][none      ][-1,-1,-1]
> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:831 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  831  ][none      ][-1,-1,-1]
> interface_id=1
> globals.SKYPIAX_INTERFACES[interface_id].skype_user=idapted_voip_11
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:835 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  835  ][none      ][-1,-1,-1]
> interface_id=1
> globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:839 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  839  ][none      ][-1,-1,-1]
> interface_id=1
> globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:842 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  842  ][none      ][-1,-1,-1]
> interface_id=1
> globals.SKYPIAX_INTERFACES[interface_id].name=idapted_voip_1
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:845 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  845  ][none      ][-1,-1,-1]
> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:849 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  849  ][none      ][-1,-1,-1]
> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:853 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  853  ][none      ][-1,-1,-1]
> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].destination=5000
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:856 load_config() rev
> 11066M[(nil)|37     ][DEBUG_SKYPE  856  ][none      ][-1,-1,-1]
> interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default
> 2009-01-15 17:56:41 [NOTICE] mod_skypiax.c:857 load_config() rev
> 11066M[(nil)|37     ][NOTICA  857  ][none      ][-1,-1,-1] STARTING
> interface_id=1
> 2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:925
> skypiax_skypeapi_thread_func() rev 11066M[(nil)|37     ][DEBUG_PBX
> 925  ][idapted_voip_1][-1, 0, 0] ENTERING FUNC
> 2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:938
> skypiax_skypeapi_thread_func() rev 11066M[(nil)|37     ][DEBUG_SKYPE
> 938  ][idapted_voip_1][-1, 0, 0] X Display ':101' opened
> 2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:890
> skypiax_skype_present() rev 11066M[(nil)|37     ][DEBUG_SKYPE  890  ]
> [none      ][-1,-1,-1] Skype instance found with id #2097250
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:570
> skypiax_signaling_thread_func() rev 11066M[(nil)|37     ][DEBUG_PBX
> 570  ][idapted_voip_1][-1, 0, 0] ENTERING FUNC
> 2009-01-15 17:56:41 [DEBUG] mod_skypiax.c:573
> skypiax_signaling_thread_func() rev 11066M[(nil)|37     ][DEBUG_SKYPE
> 573  ][idapted_voip_1][-1, 0, 0] In skypiax_signaling_thread_func:
> started, p=0xa792a420
> 2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147
> skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ]
> [idapted_voip_1][-1, 0, 0] READING: |||OK|||
> 2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147
> skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ]
> [idapted_voip_1][-1, 0, 0] READING: |||PROTOCOL 6|||
> 2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147
> skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ]
> [idapted_voip_1][-1, 0, 0] READING: |||CONNSTATUS ONLINE|||
> 2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147
> skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ]
> [idapted_voip_1][-1, 0, 0] READING: |||CURRENTUSERHANDLE
> idapted_voip_1|2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147
> skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ]
> [idapted_voip_1][-1, 0, 0] READING: |||CURRENTUSERHANDLE
> idapted_voip_1|||
> 2009-01-15 17:56:41 [DEBUG] skypiax_protocol.c:1147
> skypiax_skype_read() rev 11066M[(nil)|37     ][DEBUG_SKYPE  1147 ]
> [idapted_voip_1][-1, 0, 0] READING: |||USERSTATUS ONLINE|||
> 2009-01-15 17:56:42 [DEBUG] skypiax_protocol.c:471
> skypiax_skypeaudio_init() rev 11066M[(nil)|37     ][DEBUG_PBX  471  ]
> [idapted_voip_1][-1, 0, 0] EXITING FUNC
>
>
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From krice at suspicious.org  Thu Jan 15 09:15:28 2009
From: krice at suspicious.org (Ken Rice)
Date: Thu, 15 Jan 2009 11:15:28 -0600
Subject: [Freeswitch-dev] [Freeswitch-users] VMWare voice quality
In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl>
Message-ID: <C594CB50.4BFC4%krice@suspicious.org>

On 1/15/09 11:01 AM, "Remko Kloosterman" <R.Kloosterman at mtel.nl> wrote:

> Hello Ken, hello all,
> 
> I just read about the FreeSWITCH VMware applicance. I'm curious about
> your experiences with the audio quality on VMWare, so here's a new
> thread. 
> 
> I've installed freeswitch on VMware Server for Windows. The IVR audio
> always plays choppy, while the server itself has no performance issues.
> The same poor voice quality also goes for Asterisk or Yate, even on a
> very fast VMware ESX system.
> 
> Did you experience the same and/or do you have pointers on how to
> troubleshoot and fix this?


There is a high resolution timer you need to enable on vmware... I'm not
familiar enuff with all the versions of vmware to advise there that switch
is, but they have a couple of articles on it in their knowledge base




From msc at freeswitch.org  Thu Jan 15 09:18:18 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 15 Jan 2009 09:18:18 -0800
Subject: [Freeswitch-dev] bridge application : how to retrieve data field
In-Reply-To: <496D9304.8030804@bewan.com>
References: <496D9304.8030804@bewan.com>
Message-ID: <87f2f3b90901150918n48f3558dm66e756f0499ae234@mail.gmail.com>

On Tue, Jan 13, 2009 at 11:23 PM, ludovic <ludovic.fouquet at bewan.com> wrote:
> Hi,
>
> I develop my own endpoint.
> I have a question regarding the xml dialplan rule like  ( I take openzap
> for the example) :
>
> <action application="bridge" data="openzap/1/1"/>
>
> I understand that the first field of data is the mod name. The remaining
> "1/1" is useful for openzap to select the right channel,
> Where is 1/1 passed in Freeswitch ? In a callback routine of the mod ?
> or elsewhere ?
>

See libs/openpzap/mod_openzap/mod_openzap.c, function name
'channel_outgoing_channel'
The dialstring gets parsed right around line 922

-MC
> Regards,
>
> Ludovic
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From gmaruzz at celliax.org  Thu Jan 15 09:39:36 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 15 Jan 2009 18:39:36 +0100
Subject: [Freeswitch-dev] skypiax (skype compatible) and celliax (gsm
	network) development plans
Message-ID: <7b197bef0901150939l3f8ee319k6cf2d6d0ee21241d@mail.gmail.com>

Hi FreeSWITCH developers

I would like to propose to the community my plans, so we can discuss
and coordinate efforts.

I developed a couple of channel drivers for Asterisk in the past
(works on both Linux and Windows), and I would like to port them to FS
and further enhance them.

The two endpoints are:
- Skypiax, Skype compatible, makes and receives calls to/from Skype
network and Skypeout service, using the Skype client as interface.
- Celliax, GSM and SMS endpoint, makes and receives voice calls and
SMSs to/from the GSM/CDMA network, using second hand cellphones and/or
embedded professional devices as interfaces

My aims are:

a) port both endpoints from Asterisk to FreeSWITCH
b) have both endpoints continue to support at least Linux and Windows on FS
c) I would like better having most of the endpoints code working for
both FreeSWITCH and Asterisk, maintaining separated the code that
interface with the GSM and Skype network, from the code that interface
with the core.

Skypiax, the skype compatible endpoint, is a fork of celliax, the GSM
endpoint, and they share the same skeleton and logic, so porting
celliax after having ported skypiax will be easier and faster :-).

Current situation and next steps:
1) skypiax (http://wiki.freeswitch.org/wiki/Skypiax) is now available
for testing and debugging, needs to be polished and cleaned
2) starting mid next week (I'll be back in office), I want to
integrate into skypiax the code and ideas from mod_airpe of Massimo
(ctrix), that has developed an alternative Skype compatible module,
and coordinate any future development with him and any other
interested developer
3) begin the porting of celliax to FS, aiming at a pre-beta release
for Linux and Windows during February.
4) coordinate further development of celliax with any other developer
interested in GSM, SMSs, CDMA, IDEN, AT commands, FBUS commands,
embedded devices, audio sampling

I am gmaruzz on #freeswitch and #freeswitch-dev, you can find more
info at www.celliax.org.

Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039


From damin at nacs.net  Thu Jan 15 12:12:34 2009
From: damin at nacs.net (Gregory Boehnlein)
Date: Thu, 15 Jan 2009 15:12:34 -0500
Subject: [Freeswitch-dev] [Freeswitch-users] VMWare voice quality
In-Reply-To: <C594CB50.4BFC4%krice@suspicious.org>
References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl>
	<C594CB50.4BFC4%krice@suspicious.org>
Message-ID: <006d01c9774d$9a94cd00$cfbe6700$@net>

That won't eliminate the problem. Just reduce the possibility of it
happening.

Trust me... I've got a large ESX infrastructure, and there is no way that a
software based Voice platform is going to provide skip free audio in a
virtualized environment.

> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-
> dev-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
> Sent: Thursday, January 15, 2009 12:15 PM
> To: freeswitch-users at lists.freeswitch.org; Remko Kloosterman;
> freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] [Freeswitch-users] VMWare voice quality
> 
> On 1/15/09 11:01 AM, "Remko Kloosterman" <R.Kloosterman at mtel.nl> wrote:
> 
> > Hello Ken, hello all,
> >
> > I just read about the FreeSWITCH VMware applicance. I'm curious about
> > your experiences with the audio quality on VMWare, so here's a new
> > thread.
> >
> > I've installed freeswitch on VMware Server for Windows. The IVR audio
> > always plays choppy, while the server itself has no performance
> issues.
> > The same poor voice quality also goes for Asterisk or Yate, even on a
> > very fast VMware ESX system.
> >
> > Did you experience the same and/or do you have pointers on how to
> > troubleshoot and fix this?
> 
> 
> There is a high resolution timer you need to enable on vmware... I'm
> not
> familiar enuff with all the versions of vmware to advise there that
> switch
> is, but they have a couple of articles on it in their knowledge base
> 
> 
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 
> --
> This message has been scanned for viruses and
> dangerous content by N2Net Mailshield, and is
> believed to be clean.



From mike at jerris.com  Thu Jan 15 12:30:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Jan 2009 15:30:26 -0500
Subject: [Freeswitch-dev] [Freeswitch-users]   VMWare voice quality
In-Reply-To: <006d01c9774d$9a94cd00$cfbe6700$@net>
References: <11372C8B9E603F4FACDE6AB18256DEC601479A79@srvmtel.office.mtel.nl>
	<C594CB50.4BFC4%krice@suspicious.org>
	<006d01c9774d$9a94cd00$cfbe6700$@net>
Message-ID: <76E232D0-2CC8-46FF-8300-A291DA0EB7AF@jerris.com>

To the contrary, we have had quite good results in virtualized  
environments and you don't really need timing that is that accurate to  
make it work.  We work quite well on amazon EC2 for example.  There  
are 2 issues I know about with vmware, 1 is you need to set a setting  
on the host to extend somewhat sane clocks being available, the second  
is I have seen issues with the bridged network adapter actually  
doubling up all packets causing very strange issues, I suggest not  
using bridged networking if you experience this.

Mike

On Jan 15, 2009, at 3:12 PM, Gregory Boehnlein wrote:

> That won't eliminate the problem. Just reduce the possibility of it
> happening.
>
> Trust me... I've got a large ESX infrastructure, and there is no way  
> that a
> software based Voice platform is going to provide skip free audio in a
> virtualized environment.
>
>> -----Original Message-----
>> From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-
>> dev-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
>> Sent: Thursday, January 15, 2009 12:15 PM
>> To: freeswitch-users at lists.freeswitch.org; Remko Kloosterman;
>> freeswitch-dev at lists.freeswitch.org
>> Subject: Re: [Freeswitch-dev] [Freeswitch-users] VMWare voice quality
>>
>> On 1/15/09 11:01 AM, "Remko Kloosterman" <R.Kloosterman at mtel.nl>  
>> wrote:
>>
>>> Hello Ken, hello all,
>>>
>>> I just read about the FreeSWITCH VMware applicance. I'm curious  
>>> about
>>> your experiences with the audio quality on VMWare, so here's a new
>>> thread.
>>>
>>> I've installed freeswitch on VMware Server for Windows. The IVR  
>>> audio
>>> always plays choppy, while the server itself has no performance
>> issues.
>>> The same poor voice quality also goes for Asterisk or Yate, even  
>>> on a
>>> very fast VMware ESX system.
>>>
>>> Did you experience the same and/or do you have pointers on how to
>>> troubleshoot and fix this?
>>
>>
>> There is a high resolution timer you need to enable on vmware... I'm
>> not
>> familiar enuff with all the versions of vmware to advise there that
>> switch
>> is, but they have a couple of articles on it in their knowledge base
>>
>>


From krice at suspicious.org  Thu Jan 15 13:01:45 2009
From: krice at suspicious.org (Ken Rice)
Date: Thu, 15 Jan 2009 15:01:45 -0600
Subject: [Freeswitch-dev] [Freeswitch-users] VMWare voice quality
In-Reply-To: <006d01c9774d$9a94cd00$cfbe6700$@net>
Message-ID: <C5950059.4C090%krice@suspicious.org>

Ok if can summarize a little of the intention of releasing this VMWare
image. Its really there so you guys can get it and check it out. I
personally don't believe in running such services on a virtual machine (too
many nightmare stories from the 'day job' from such things)

However, for testing and developing applications that ride on top of
FreeSWITCH, this is a quick way to get up and running.

Remember Voice application especially where you are interacting with the
media streams will be affected by latency and jitter much more readily then
store and forward things like IRC, Web, eMail and instant messaging.

K


On 1/15/09 2:12 PM, "Gregory Boehnlein" <damin at nacs.net> wrote:

> That won't eliminate the problem. Just reduce the possibility of it
> happening.
> 
> Trust me... I've got a large ESX infrastructure, and there is no way that a
> software based Voice platform is going to provide skip free audio in a
> virtualized environment.




From anthony.minessale at gmail.com  Thu Jan 15 15:40:29 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Jan 2009 17:40:29 -0600
Subject: [Freeswitch-dev] Using ODBC core from FreeSWITCH module
In-Reply-To: <03AFD2B5818449BF8C780A9719BCD6D8@test>
References: <Acl22Et4ShKIAxGGTnC+tD+w8ySfng==>
	<03AFD2B5818449BF8C780A9719BCD6D8@test>
Message-ID: <191c3a030901151540p2ffc5a6bvb01658c2559fc4e0@mail.gmail.com>

try trunk i think i found your issue and fixed it.


On Thu, Jan 15, 2009 at 12:12 AM, Darren Schreiber <d at d-man.org> wrote:

>  Hey there,
>     I'm using the ODBC abilities of FreeSWITCH for my billing module and I
> am having an issue.
>
> When calling:
>
>                 if (!(switch_odbc_handle_callback_exec(globals.master_odbc,
> SQL_UPDATE_STATEMENT, nibblebill_callback, &pdata) == SWITCH_ODBC_SUCCESS)){
> sometimes the call is actually failing but I am always getting an ODBC
> success. I am thinking "failing" is really 0 rows updated. How should I be
> checking for this occurrence?
>
> - Darren
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From seven at idapted.com  Thu Jan 15 21:54:43 2009
From: seven at idapted.com (seven du)
Date: Fri, 16 Jan 2009 13:54:43 +0800
Subject: [Freeswitch-dev] mod_skypiax inching forward
In-Reply-To: <mailman.3707.1232051449.18013.freeswitch-dev@lists.freeswitch.org>
References: <mailman.3707.1232051449.18013.freeswitch-dev@lists.freeswitch.org>
Message-ID: <A6C779B5-37D9-450E-BD31-261A6BECC578@idapted.com>

Hi Giovanni,




root at db1:~/configs# psgrep free
root      3962  2.6  1.4  16324 11708 pts/5    S    13:21   0:00 /usr/ 
bin/Xvfb :101 -auth /usr/local/freeswitch/conf/autoload_configs/ 
skypiax.X.conf
root      3976  0.0  0.0   1692   504 pts/5    R+   13:21   0:00 grep  
free
root at db1:~/configs# psgrep skype
root      3967 16.0  3.7  56708 29152 pts/5    Sl   13:21   0:00 /usr/ 
bin/skype --pipelogin
root      3978  0.0  0.0   1692   504 pts/5    R+   13:21   0:00 grep  
skype
root at db1:~/configs# lsmod|grep snd
snd_dummy              13568  0
snd_pcm                75780  1 snd_dummy
snd_timer              24708  1 snd_pcm
snd                    55172  3 snd_dummy,snd_pcm,snd_timer
soundcore               8800  1 snd
snd_page_alloc         11400  1 snd_pcm
root at db1:~/configs#


The  console stuck when loading mod_skypiax, however, other debug  
messages( like register to vitelity)  still being printed out. When I  
call in using another skype account, freeswitch crashed.


freeswitch at db1.veecue.com>
freeswitch at db1.veecue.com> console loglevel 9
API CALL [console(loglevel 9)] output:
+OK console log level set to DEBUG

freeswitch at db1.veecue.com> load mod_skypiax
2009-01-16 13:47:49 [ERR] mod_xml_curl.c:230 xml_url_fetch() Received  
HTTP error 404 trying to fetch http://fs.lan:80/voip_configurations/show
data:  
[hostname 
= 
db1 
.veecue 
.com 
&section 
= 
configuration 
&tag_name=configuration&key_name=name&key_value=skypiax.conf]
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:630 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  630  ][none      ][-1,-1,-1]  
globals.debug=0
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:632 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  632  ][none      ][-1,-1,-1]  
globals.debug=8
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:643 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  643  ][none      ][-1,-1,-1] codec- 
master globals.debug=8
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:646 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  646  ][none      ][-1,-1,-1]  
globals.dialplan=XML
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:652 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  652  ][none      ][-1,-1,-1]  
globals.context=default
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:649 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  649  ][none      ][-1,-1,-1]  
globals.destination=5000
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:655 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  655  ][none      ][-1,-1,-1]  
globals.codec_string=gsm,ulaw
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:662 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  662  ][none      ][-1,-1,-1]  
globals.codec_rates_string=8000,16000
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:635 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  635  ][none      ][-1,-1,-1]  
globals.hold_music=local_stream://moh
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:757 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  757  ][none      ][-1,-1,-1]  
interface_id=1
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:780 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  780  ][none      ][-1,-1,-1]  
name=idapted_voip_1
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:786 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  786  ][none      ][-1,-1,-1]  
Initialized XInitThreads!
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:798 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  798  ][none      ][-1,-1,-1]  
CONFIGURING interface_id=1
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:827 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  827  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:831 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  831  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].skype_user=idapted_voip_11
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:835 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  835  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:839 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  839  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:842 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  842  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].name=idapted_voip_1
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:845 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  845  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:849 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  849  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:853 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  853  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].destination=5000
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:856 load_config() rev  
11253M[(nil)|37     ][DEBUG_SKYPE  856  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default
2009-01-16 13:47:49 [NOTICE] mod_skypiax.c:857 load_config() rev  
11253M[(nil)|37     ][NOTICA  857  ][none      ][-1,-1,-1] STARTING  
interface_id=1
2009-01-16 13:47:49 [DEBUG] skypiax_protocol.c:925  
skypiax_skypeapi_thread_func() rev 11253M[(nil)|37     ][DEBUG_PBX   
925  ][idapted_voip_1][-1, 0, 0] ENTERING FUNC
2009-01-16 13:47:49 [DEBUG] skypiax_protocol.c:938  
skypiax_skypeapi_thread_func() rev 11253M[(nil)|37     ][DEBUG_SKYPE   
938  ][idapted_voip_1][-1, 0, 0] X Display ':101' opened
2009-01-16 13:47:49 [DEBUG] skypiax_protocol.c:890  
skypiax_skype_present() rev 11253M[(nil)|37     ][DEBUG_SKYPE  890  ] 
[none      ][-1,-1,-1] Skype instance found with id #2097250
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:570  
skypiax_signaling_thread_func() rev 11253M[(nil)|37     ][DEBUG_PBX   
570  ][idapted_voip_1][-1, 0, 0] ENTERING FUNC
2009-01-16 13:47:49 [DEBUG] mod_skypiax.c:573  
skypiax_signaling_thread_func() rev 11253M[(nil)|37     ][DEBUG_SKYPE   
573  ][idapted_voip_1][-1, 0, 0] In skypiax_signaling_thread_func:  
started, p=0xa7908420
2009-01-16 13:47:49 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||OK|||
2009-01-16 13:47:49 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||PROTOCOL 6|||
2009-01-16 13:47:49 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||CONNSTATUS ONLINE|||
2009-01-16 13:47:49 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||CURRENTUSERHANDLE  
idapted_voip_1|||
2009-01-16 13:47:49 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||USERSTATUS AWAY|||
2009-01-16 13:47:50 [DEBUG] skypiax_protocol.c:471  
skypiax_skypeaudio_init() rev 11253M[(nil)|37     ][DEBUG_PBX  471  ] 
[idapted_voip_1][-1, 0, 0] EXITING FUNC







2009-01-16 13:48:34 [NOTICE] sofia_reg.c:265 sofia_reg_check_gateway()  
Registering vitelity
2009-01-16 13:48:34 [DEBUG] sofia.c:462 sofia_event_callback()  
nua_i_outbound: unknown event 8: 101 NAT detected
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||CALL 39 CONF_ID 0|||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1338  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1338 ] 
[idapted_voip_1][-1, 0, 0] the skype_call 39 is NOT a conference call
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||CALL 39 STATUS RINGING|||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1039  
skypiax_skype_write() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1039 ] 
[idapted_voip_1][-1, 2,114] SENDING: |||SET AGC OFF||||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1039  
skypiax_skype_write() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1039 ] 
[idapted_voip_1][-1, 2,114] SENDING: |||SET AEC OFF||||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1039  
skypiax_skype_write() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1039 ] 
[idapted_voip_1][-1, 2,114] SENDING: |||GET CALL 39 PARTNER_DISPNAME||||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1039  
skypiax_skype_write() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1039 ] 
[idapted_voip_1][-1, 2,114] SENDING: |||GET CALL 39 PARTNER_HANDLE||||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1039  
skypiax_skype_write() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1039 ] 
[idapted_voip_1][-1, 2,114] SENDING: |||ALTER CALL 39 ANSWER||||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1411  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1411 ] 
[idapted_voip_1][-1, 2,114] We answered a Skype RING on skype_call 39
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 2,114] READING: ||||||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 2,114] READING: ||||||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 2,114] READING: |||CALL 39 PARTNER_DISPNAME  
seven1240|||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1334  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1334 ] 
[idapted_voip_1][-1, 2,114] the skype_call 39 caller PARTNER_DISPNAME  
(tech_pvt->callid_name) is: seven1240
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 2,114] READING: |||CALL 39 PARTNER_HANDLE  
seven1240|||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1326  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1326 ] 
[idapted_voip_1][-1, 2,114] the skype_call 39 caller PARTNER_HANDLE  
(tech_pvt->callid_number) is: seven1240
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 2,114] READING: |||AGC OFF|||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 2,114] READING: |||AEC OFF|||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 2,114] READING: |||CALL 39 STATUS INPROGRESS|||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1525  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1525 ] 
[idapted_voip_1][-1, 5,115] skype_call: 39 is now active
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1529  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1529 ] 
[idapted_voip_1][-1, 5,115] skype_call: 39 SKYPIAX_CONTROL_ANSWER sent
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:67  
skypiax_do_tcp_srv_thread() rev 11253M[(nil)|37     ][DEBUG_SKYPE   
67   ][idapted_voip_1][-1, 5,115] started tcp_srv_thread thread. 
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1541  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1541 ] 
[idapted_voip_1][-1, 5,115] started tcp_srv_thread thread.
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:320  
skypiax_do_tcp_cli_thread() rev 11253M[(nil)|37     ][DEBUG_SKYPE   
320  ][idapted_voip_1][-1, 5,115] started tcp_cli_thread thread.
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1549  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1549 ] 
[idapted_voip_1][-1, 5,115] started tcp_cli_thread thread.
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1039  
skypiax_skype_write() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1039 ] 
[idapted_voip_1][-1, 5,115] SENDING: |||ALTER CALL 39 SET_OUTPUT  
PORT="15557"||||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:73  
skypiax_do_tcp_srv_thread() rev 11253M[(nil)|37     ][DEBUG_SKYPE   
73   ][idapted_voip_1][-1, 5,115] ACCEPTED
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1039  
skypiax_skype_write() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1039 ] 
[idapted_voip_1][-1, 5,115] SENDING: |||ALTER CALL 39 SET_INPUT  
PORT="15556"||||
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:326  
skypiax_do_tcp_cli_thread() rev 11253M[(nil)|37     ][DEBUG_SKYPE   
326  ][idapted_voip_1][-1, 5,115] ACCEPTED
2009-01-16 13:48:52 [DEBUG] skypiax_protocol.c:1572  
skypiax_skype_read() rev 11253M[(nil)|37     ][DEBUG_SKYPE  1572 ] 
[idapted_voip_1][-1, 5, 5] New Inbound Channel!
2009-01-16 13:48:52 [CRIT] switch_core_session.c:1048  
switch_core_session_request_uuid() The system cannot create any  
sessions at this time.
freeswitch: src/switch_channel.c:1705:  
switch_channel_perform_mark_answered: Assertion `channel != ((void  
*)0)' failed.
Aborted (core dumped)
root at db1:/usr/local/freeswitch#
root at db1:/usr/local/freeswitch#


<configuration name="skypiax.conf" description="Skypiax Configuration">
   <global_settings>
     <param name="debug" value="8"/>
     <param name="codec-master" value="us"/>
     <param name="dialplan" value="XML"/>
     <param name="context" value="default"/>
     <param name="destination" value="5000"/>
     <param name="codec-prefs" value="gsm,ulaw"/>
     <param name="codec-rates" value="8000,16000"/>
     <param name="hold-music" value="$${moh_uri}"/>
   </global_settings>
   <!-- one entry here per skypiax interface -->
   <per_interface_settings>
     <interface id="1" name="idapted_voip_1">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":101"/>
         <param name="tcp_cli_port" value="15556"/>
         <param name="tcp_srv_port" value="15557"/>
         <param name="skype_user" value="idapted_voip_11"/>
     </interface>
     <!-- following interfaces are commented out!
     <interface id="2" name="idapted_voip_2">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":102"/>
         <param name="tcp_cli_port" value="15558"/>
         <param name="tcp_srv_port" value="15559"/>
         <param name="skype_user" value="idapted_voip_2"/>
     </interface>
     <interface id="3" name="skypiax3">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":103"/>
         <param name="tcp_cli_port" value="15560"/>
         <param name="tcp_srv_port" value="15561"/>
         <param name="skype_user" value="skypiax3"/>
     </interface>

     <interface id="4" name="skypiax4">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":104"/>
         <param name="tcp_cli_port" value="15562"/>
         <param name="tcp_srv_port" value="15563"/>
         <param name="skype_user" value="skypiax4"/>
     </interface>

     <interface id="5" name="skypiax5">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":105"/>
         <param name="tcp_cli_port" value="15564"/>
         <param name="tcp_srv_port" value="15565"/>
         <param name="skype_user" value="skypiax5"/>
     </interface>
     <interface id="6" name="skypiax6">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":106"/>
         <param name="tcp_cli_port" value="15566"/>
         <param name="tcp_srv_port" value="15567"/>
         <param name="skype_user" value="skypiax6"/>
     </interface>
     <interface id="7" name="skypiax7">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":107"/>
         <param name="tcp_cli_port" value="15568"/>
         <param name="tcp_srv_port" value="15569"/>
         <param name="skype_user" value="skypiax7"/>
     </interface>

     <interface id="8" name="skypiax8">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":108"/>
         <param name="tcp_cli_port" value="15570"/>
         <param name="tcp_srv_port" value="15571"/>
         <param name="skype_user" value="skypiax8"/>
     </interface>

     <interface id="9" name="skypiax9">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":109"/>
         <param name="tcp_cli_port" value="15572"/>
         <param name="tcp_srv_port" value="15573"/>
         <param name="skype_user" value="skypiax9"/>
     </interface>

     <interface id="10" name="skypiax10">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":110"/>
         <param name="tcp_cli_port" value="15574"/>
         <param name="tcp_srv_port" value="15575"/>
         <param name="skype_user" value="skypiax10"/>
     </interface>

     <interface id="11" name="skypiax11">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":111"/>
         <param name="tcp_cli_port" value="15576"/>
         <param name="tcp_srv_port" value="15577"/>
         <param name="skype_user" value="skypiax11"/>
     </interface>

     <interface id="12" name="skypiax12">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":112"/>
         <param name="tcp_cli_port" value="15578"/>
         <param name="tcp_srv_port" value="15579"/>
         <param name="skype_user" value="skypiax12"/>
     </interface>

     <interface id="13" name="skypiax13">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":113"/>
         <param name="tcp_cli_port" value="15580"/>
         <param name="tcp_srv_port" value="15581"/>
         <param name="skype_user" value="skypiax13"/>
     </interface>

     <interface id="14" name="skypiax14">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":114"/>
         <param name="tcp_cli_port" value="15582"/>
         <param name="tcp_srv_port" value="15583"/>
         <param name="skype_user" value="skypiax14"/>
     </interface>

     <interface id="15" name="skypiax15">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":115"/>
         <param name="tcp_cli_port" value="15584"/>
         <param name="tcp_srv_port" value="15585"/>
         <param name="skype_user" value="skypiax15"/>
     </interface>

     <interface id="16" name="skypiax16">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":116"/>
         <param name="tcp_cli_port" value="15586"/>
         <param name="tcp_srv_port" value="15587"/>
         <param name="skype_user" value="skypiax16"/>
     </interface>

     <interface id="17" name="skypiax17">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":117"/>
         <param name="tcp_cli_port" value="15588"/>
         <param name="tcp_srv_port" value="15589"/>
         <param name="skype_user" value="skypiax17"/>
     </interface>

     <interface id="18" name="skypiax18">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":118"/>
         <param name="tcp_cli_port" value="15590"/>
         <param name="tcp_srv_port" value="15591"/>
         <param name="skype_user" value="skypiax18"/>
     </interface>

     <interface id="19" name="skypiax19">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":119"/>
         <param name="tcp_cli_port" value="15592"/>
         <param name="tcp_srv_port" value="15593"/>
         <param name="skype_user" value="skypiax19"/>
     </interface>

     <interface id="20" name="skypiax20">
         <param name="hold-music" value="$${moh_uri}"/>
         <param name="dialplan" value="XML"/>
         <param name="context" value="default"/>
         <param name="X11-display" value=":120"/>
         <param name="tcp_cli_port" value="15594"/>
         <param name="tcp_srv_port" value="15595"/>
         <param name="skype_user" value="skypiax20"/>
     </interface>
     -->
   </per_interface_settings>
</configuration>


btw, I can only receive the digest, how can I receive mails  
seperately? so I can reply more easier.


-------
Hi Seven,
Seems that there are no errors, and I don't think is a sound problem,
at the moment.

Also, your Skype config.xml seems correct.

I see you have an old version, could you please svn update the skypiax
code, recompile, and try again?

If there still problems, can you post the debug after "console
loglevel 9", the skypiax.conf.xml, and the lsmod|grep snd?

Ciao for now,

Giovanni 
                                 
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From seven at idapted.com  Fri Jan 16 01:02:08 2009
From: seven at idapted.com (seven du)
Date: Fri, 16 Jan 2009 17:02:08 +0800
Subject: [Freeswitch-dev]  mod_skypiax inching forward
Message-ID: <5025FE6C-A72F-475A-9D8F-0F44DD59C3A9@idapted.com>

hi Giovanni,

I successfully loaded skypiax on another ubuntu gutsy machine, the  
privious problem maybe because it is in a Xen VM.

But, when I call out with originate skype/echo123 &echo, it shows  
chan_not_implemented, however, I think it should be skypiax(2009-01-16  
16:49:06 [NOTICE] switch_loadable_module.c:141  
switch_loadable_module_process() Adding Endpoint 'skypiax'). when I  
dial with originate skypiax/echo123 &echo, core dumped.

freeswitch at djf-desktop> originate skype/echo123 &echo
2009-01-16 16:53:36 [ERR] switch_core_session.c:255  
switch_core_session_outgoing_channel() Could not locate channel type  
skype
2009-01-16 16:53:36 [ERR] switch_ivr_originate.c:1122  
switch_ivr_originate() Cannot create outgoing channel of type [skype]  
cause: [CHAN_NOT_IMPLEMENTED]
2009-01-16 16:53:36 [DEBUG] switch_ivr_originate.c:1705  
switch_ivr_originate() Originate Resulted in Error Cause: 66  
[CHAN_NOT_IMPLEMENTED]
API CALL [originate(skype/echo123 &echo)] output:
-ERR CHAN_NOT_IMPLEMENTED

freeswitch at djf-desktop>
freeswitch at djf-desktop> console loglevel 9
API CALL [console(loglevel 9)] output:
+OK console log level set to DEBUG

freeswitch at djf-desktop> load mod_skypiax
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:630 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  630  ][none      ][-1,-1,-1]  
globals.debug=0
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:632 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  632  ][none      ][-1,-1,-1]  
globals.debug=8
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:643 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  643  ][none      ][-1,-1,-1] codec- 
master globals.debug=8
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:646 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  646  ][none      ][-1,-1,-1]  
globals.dialplan=XML
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:652 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  652  ][none      ][-1,-1,-1]  
globals.context=default
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:649 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  649  ][none      ][-1,-1,-1]  
globals.destination=5000
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:655 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  655  ][none      ][-1,-1,-1]  
globals.codec_string=gsm,ulaw
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:662 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  662  ][none      ][-1,-1,-1]  
globals.codec_rates_string=8000,16000
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:635 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  635  ][none      ][-1,-1,-1]  
globals.hold_music=
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:757 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  757  ][none      ][-1,-1,-1]  
interface_id=1
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:780 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  780  ][none      ][-1,-1,-1]  
name=idapted_voip_1
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:786 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  786  ][none      ][-1,-1,-1]  
Initialized XInitThreads!
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:798 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  798  ][none      ][-1,-1,-1]  
CONFIGURING interface_id=1
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:827 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  827  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].X11_display=:101
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:831 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  831  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].skype_user=idapted_voip_1
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:835 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  835  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].tcp_cli_port=15556
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:839 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  839  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].tcp_srv_port=15557
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:842 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  842  ][none      ][-1,-1,-1]  
interface_id=1  
globals.SKYPIAX_INTERFACES[interface_id].name=idapted_voip_1
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:845 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  845  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:849 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  849  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].dialplan=XML
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:853 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  853  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].destination=5000
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:856 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  856  ][none      ][-1,-1,-1]  
interface_id=1 globals.SKYPIAX_INTERFACES[interface_id].context=default
2009-01-16 16:49:05 [NOTICE] mod_skypiax.c:857 load_config() rev  
11253[(nil)|37     ][NOTICA  857  ][none      ][-1,-1,-1] STARTING  
interface_id=1
2009-01-16 16:49:05 [DEBUG] skypiax_protocol.c:925  
skypiax_skypeapi_thread_func() rev 11253[(nil)|37     ][DEBUG_PBX   
925  ][idapted_voip_1][-1, 0, 0] ENTERING FUNC
2009-01-16 16:49:05 [DEBUG] skypiax_protocol.c:938  
skypiax_skypeapi_thread_func() rev 11253[(nil)|37     ][DEBUG_SKYPE   
938  ][idapted_voip_1][-1, 0, 0] X Display ':101' opened
2009-01-16 16:49:05 [DEBUG] skypiax_protocol.c:890  
skypiax_skype_present() rev 11253[(nil)|37     ][DEBUG_SKYPE  890  ] 
[none      ][-1,-1,-1] Skype instance found with id #2097367
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:570  
skypiax_signaling_thread_func() rev 11253[(nil)|37     ][DEBUG_PBX   
570  ][idapted_voip_1][-1, 0, 0] ENTERING FUNC
2009-01-16 16:49:05 [DEBUG] mod_skypiax.c:573  
skypiax_signaling_thread_func() rev 11253[(nil)|37     ][DEBUG_SKYPE   
573  ][idapted_voip_1][-1, 0, 0] In skypiax_signaling_thread_func:  
started, p=0xab31e440
2009-01-16 16:49:05 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||OK|||
2009-01-16 16:49:05 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||PROTOCOL 6|||
2009-01-16 16:49:05 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||CONNSTATUS ONLINE|||
2009-01-16 16:49:05 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||CURRENTUSERHANDLE  
idapted_voip_1|||
2009-01-16 16:49:05 [DEBUG] skypiax_protocol.c:1147  
skypiax_skype_read() rev 11253[(nil)|37     ][DEBUG_SKYPE  1147 ] 
[idapted_voip_1][-1, 0, 0] READING: |||USERSTATUS NA|||
2009-01-16 16:49:06 [DEBUG] skypiax_protocol.c:471  
skypiax_skypeaudio_init() rev 11253[(nil)|37     ][DEBUG_PBX  471  ] 
[idapted_voip_1][-1, 0, 0] EXITING FUNC
2009-01-16 16:49:06 [NOTICE] mod_skypiax.c:886 load_config() rev  
11253[(nil)|37     ][NOTICA  886  ][none      ][-1,-1,-1] STARTED  
interface_id=1
API CALL [load(mod_skypiax)] output:
+OK

2009-01-16 16:49:06 [DEBUG] mod_skypiax.c:899 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  899  ][none      ][-1,-1,-1] i=1  
globals.SKYPIAX_INTERFACES[1].interface_id=1
2009-01-16 16:49:06 [DEBUG] mod_skypiax.c:901 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  901  ][none      ][-1,-1,-1] i=1  
globals.SKYPIAX_INTERFACES[1].X11_display=:101
2009-01-16 16:49:06 [DEBUG] mod_skypiax.c:903 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  903  ][none      ][-1,-1,-1] i=1  
globals.SKYPIAX_INTERFACES[1].name=idapted_voip_1
2009-01-16 16:49:06 [DEBUG] mod_skypiax.c:905 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  905  ][none      ][-1,-1,-1] i=1  
globals.SKYPIAX_INTERFACES[1].context=default
2009-01-16 16:49:06 [DEBUG] mod_skypiax.c:907 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  907  ][none      ][-1,-1,-1] i=1  
globals.SKYPIAX_INTERFACES[1].dialplan=XML
2009-01-16 16:49:06 [DEBUG] mod_skypiax.c:909 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  909  ][none      ][-1,-1,-1] i=1  
globals.SKYPIAX_INTERFACES[1].destination=5000
2009-01-16 16:49:06 [DEBUG] mod_skypiax.c:911 load_config() rev  
11253[(nil)|37     ][DEBUG_SKYPE  911  ][none      ][-1,-1,-1] i=1  
globals.SKYPIAX_INTERFACES[1].context=default
2009-01-16 16:49:06 [DEBUG] mod_skypiax.c:938 mod_skypiax_load() rev  
11253[(nil)|37     ][DEBUG_SKYPE  938  ][none      ][-1,-1,-1] EXITING  
FUNC!
2009-01-16 16:49:06 [CONSOLE] switch_loadable_module.c:857  
switch_loadable_module_load_file() Successfully Loaded [mod_skypiax]
2009-01-16 16:49:06 [NOTICE] switch_loadable_module.c:141  
switch_loadable_module_process() Adding Endpoint 'skypiax'
freeswitch at djf-desktop>
freeswitch at djf-desktop>
freeswitch at djf-desktop> originate skype/echo123 &echo
2009-01-16 16:53:36 [ERR] switch_core_session.c:255  
switch_core_session_outgoing_channel() Could not locate channel type  
skype
2009-01-16 16:53:36 [ERR] switch_ivr_originate.c:1122  
switch_ivr_originate() Cannot create outgoing channel of type [skype]  
cause: [CHAN_NOT_IMPLEMENTED]
2009-01-16 16:53:36 [DEBUG] switch_ivr_originate.c:1705  
switch_ivr_originate() Originate Resulted in Error Cause: 66  
[CHAN_NOT_IMPLEMENTED]
API CALL [originate(skype/echo123 &echo)] output:
-ERR CHAN_NOT_IMPLEMENTED



But I can call-IN using another skype account, and it can bridge to a  
sofia endpoint , sound quality is very good.


From seven at idapted.com  Fri Jan 16 01:37:52 2009
From: seven at idapted.com (seven du)
Date: Fri, 16 Jan 2009 17:37:52 +0800
Subject: [Freeswitch-dev]  mod_skypiax inching forward
Message-ID: <FAE5E07E-BA42-4605-8F33-C3622935C118@idapted.com>

Hi Giovanni,

Sorry for the previous mail. I noticed it should be originate skypiax/ 
my_skypy_account_name/another_skypy_name, it works, great!




From seven at idapted.com  Fri Jan 16 01:57:20 2009
From: seven at idapted.com (seven du)
Date: Fri, 16 Jan 2009 17:57:20 +0800
Subject: [Freeswitch-dev]  mod_skypiax inching forward
Message-ID: <930137EB-8BE1-474F-ADAB-02CAA94960FF@idapted.com>

is there some commands like sofia status?


From gmaruzz at celliax.org  Fri Jan 16 02:12:04 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 16 Jan 2009 11:12:04 +0100
Subject: [Freeswitch-dev] mod_skypiax inching forward
In-Reply-To: <930137EB-8BE1-474F-ADAB-02CAA94960FF@idapted.com>
References: <930137EB-8BE1-474F-ADAB-02CAA94960FF@idapted.com>
Message-ID: <7b197bef0901160212h27c01119ybe7785513ed5ed81@mail.gmail.com>

On Fri, Jan 16, 2009 at 10:57 AM, seven du <seven at idapted.com> wrote:
>Hi Giovanni,
>
>Sorry for the previous mail. I noticed it should be originate skypiax/my_skypy_account_name/another_skypy_name, it works, great!
>
> is there some commands like sofia status?
>

Dear Seven,

thanks a lot for taking time to test it out.
Happy it worked for you!

At the moment there is not a status command, eg: there are no command at all.

They'll be added very soon, next week or so.

Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Fri, Jan 16, 2009 at 10:57 AM, seven du <seven at idapted.com> wrote:
> is there some commands like sofia status?
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From seven at idapted.com  Sat Jan 17 05:06:58 2009
From: seven at idapted.com (seven du)
Date: Sat, 17 Jan 2009 21:06:58 +0800
Subject: [Freeswitch-dev]  mod_skypiax inching forward
Message-ID: <AE0204A0-AC75-4D8D-BCE2-3610D8305893@idapted.com>

Hi Giovanni,

In addition to the command line tools, I suggest taking following  
things in to account:

1) originate skypiax/wrong_skype_name won't cause core dump
2) currently we can only call skypiax/skypiax1/other_skype_name,  can  
we implement
      something like openzap, so skype group can be managed, and also  
fs knows the
      channel status. So it's easy to call out with skypiax/ 
some_skypiax_group/other_skype_name.
3) What will happen if a skype client crash? can it automatically  
disable the channel
     and reset the channel after the skype client recovered?

Best-
Seven


From gmaruzz at celliax.org  Sun Jan 18 08:31:29 2009
From: gmaruzz at celliax.org (gmaruzz at celliax.org)
Date: Sun, 18 Jan 2009 17:31:29 +0100
Subject: [Freeswitch-dev] mod_skypiax inching forward
In-Reply-To: <AE0204A0-AC75-4D8D-BCE2-3610D8305893@idapted.com>
References: <AE0204A0-AC75-4D8D-BCE2-3610D8305893@idapted.com>
Message-ID: <7b197bef0901180831o31f511favb0442470093421d1@mail.gmail.com>

On 1/17/09, seven du <seven at idapted.com> wrote:
> In addition to the command line tools, I suggest taking following
> things in to account:
>
> 1) originate skypiax/wrong_skype_name won't cause core dump
> 2) currently we can only call skypiax/skypiax1/other_skype_name,  can
> we implement
>       something like openzap, so skype group can be managed, and also
> fs knows the
>       channel status. So it's easy to call out with skypiax/
> some_skypiax_group/other_skype_name.
> 3) What will happen if a skype client crash? can it automatically
> disable the channel
>      and reset the channel after the skype client recovered?

Seven,
thanks for suggestions, I'll start on them asap!

Do you have other ideas/suggestions/hints on how to make skypiax more
useful/easy to use?

-- 
Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039


From seven at idapted.com  Sun Jan 18 22:56:49 2009
From: seven at idapted.com (seven du)
Date: Mon, 19 Jan 2009 14:56:49 +0800
Subject: [Freeswitch-dev] mod_skypiax inching forward
Message-ID: <8075FDF7-F81B-4ED1-9DEC-70935F3589E3@idapted.com>

Giovanni,

Thank you for taking mu suggestions take into account.

Another idea: Make the skype in context and dialplan  
configurable(other than 5000?)
maybe useful. But not sure, due to the limits of skype in, perhaps the  
only use of
skype in is transfer into an ivr.


And, Just found a easy way to create bunch of skype configurations.

If you have a bunch of skype accounts like skypiax1, skypiax2,  
skypiax3 ...
You only have to login into skypiax1 one time and chose the right sound
device and other settings like disable events etc. and you will get a  
dir in
{your home dir}/.Skype/skypiax1. Just copy skypiax1 into skypiax2,  
skypiax3...
They will have similar configurations, as long as they share a same  
password.

Cheers-

Seven

 > Seven,
 > thanks for suggestions, I'll start on them asap!
 >
 > Do you have other ideas/suggestions/hints on how to make skypiax more
useful/easy to use?


From seven at idapted.com  Tue Jan 20 01:47:35 2009
From: seven at idapted.com (seven du)
Date: Tue, 20 Jan 2009 17:47:35 +0800
Subject: [Freeswitch-dev] Is there any plans to implement some kind of
	measures and statistics function?
Message-ID: <BD056EFB-86AE-4E6C-8020-7197511FE6AA@idapted.com>

Hi developers,

Now and then we need to know some statistics message and sure we need  
some way to measure it. My ideas just like this:

http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Extended_Reports_.28RTCP_XR.29_support

Is there some plans to build this kind of function in? If not, can  
some one tell me how easy or hard to do it? I think it would be better  
to use pcap or 3rd party stuffs the do that.

Thanks.


Seven


From seven at idapted.com  Tue Jan 20 01:58:19 2009
From: seven at idapted.com (seven du)
Date: Tue, 20 Jan 2009 17:58:19 +0800
Subject: [Freeswitch-dev] Is it possible to add a new profile configuration
	and enable it without restarting the server?
Message-ID: <75D59EE6-C68A-4322-8D67-E3E046629C99@idapted.com>

Hi,

For production use, we want the service run consistently and long as  
possible.  As we have command tools to reload xml configurations and  
restart gateways, I have a problem to make my configuration take  
effect before restarting the whole FreeSWITCH server.

Here are two questions:

1) When I add a gateway to a profile, say external, I can do reloadxml  
and sofia profile external restart. But if I add a new profile, how  
can I do it as it will complain the profile name not exist?

2) Is it possible to enable/disable/restart a single gateway?

Thanks

Seven


From jalsot at gmail.com  Tue Jan 20 02:04:00 2009
From: jalsot at gmail.com (Tamas)
Date: Tue, 20 Jan 2009 11:04:00 +0100
Subject: [Freeswitch-dev] Is there any plans to implement some kind of
 measures and statistics function?
In-Reply-To: <BD056EFB-86AE-4E6C-8020-7197511FE6AA@idapted.com>
References: <BD056EFB-86AE-4E6C-8020-7197511FE6AA@idapted.com>
Message-ID: <4975A190.8080001@gmail.com>

Hello Seven,

We are the one put the bounty :)
Do you mind to add some $$$ to the fund? That could help the thing make 
happen.

Regards,
    Tamas

ps: Right waiting for response from consulting@ for the request.

seven du ?rta:
> Hi developers,
>
> Now and then we need to know some statistics message and sure we need  
> some way to measure it. My ideas just like this:
>
> http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Extended_Reports_.28RTCP_XR.29_support
>
> Is there some plans to build this kind of function in? If not, can  
> some one tell me how easy or hard to do it? I think it would be better  
> to use pcap or 3rd party stuffs the do that.
>
> Thanks.
>
>
> Seven
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>   



From brian at freeswitch.org  Tue Jan 20 03:37:42 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 20 Jan 2009 05:37:42 -0600
Subject: [Freeswitch-dev] Is it possible to add a new profile
	configuration and enable it without restarting the server?
In-Reply-To: <75D59EE6-C68A-4322-8D67-E3E046629C99@idapted.com>
References: <75D59EE6-C68A-4322-8D67-E3E046629C99@idapted.com>
Message-ID: <C93274EC-305A-44D1-94DB-25C314326C8B@freeswitch.org>


On Jan 20, 2009, at 3:58 AM, seven du wrote:

> Hi,
>
> For production use, we want the service run consistently and long as
> possible.  As we have command tools to reload xml configurations and
> restart gateways, I have a problem to make my configuration take
> effect before restarting the whole FreeSWITCH server.
>
> Here are two questions:
>
> 1) When I add a gateway to a profile, say external, I can do reloadxml
> and sofia profile external restart. But if I add a new profile, how
> can I do it as it will complain the profile name not exist?

http://wiki.freeswitch.org/wiki/Sofia#Reloading_profiles_and_gateways
http://wiki.freeswitch.org/wiki/Sofia#Deleting_gateways

If you add a profile its simple.

sofia profile XXX start

/b

>
>
> 2) Is it possible to enable/disable/restart a single gateway?
>
> Thanks
>
> Seven



From jkr888 at gmail.com  Tue Jan 20 10:43:51 2009
From: jkr888 at gmail.com (Johny Kadarisman)
Date: Tue, 20 Jan 2009 13:43:51 -0500
Subject: [Freeswitch-dev] Latest build - Socket inbound variable_* on custom
	event problem.
Message-ID: <b0a271780901201043te516f6dm5e3699e7e4d160c4@mail.gmail.com>

Hi all,

Is there a way to propagate variable_* into custom event. I'm
connecting with inbound socket, and listening to that custom events.

during recent trunk update, I encountered an issues from external
application, after look around, found out that these value is not in
the events. After search around, found out some reference about
"verbose_events" with outbound sockets. So, I insert that applications
in dialplan, but still don't see any variable_* being propagate.

Is there anyway to enable this for inbound sockets?

Thanks,
Johny K.


From anthony.minessale at gmail.com  Tue Jan 20 10:54:50 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 20 Jan 2009 12:54:50 -0600
Subject: [Freeswitch-dev] Latest build - Socket inbound variable_* on
	custom event problem.
In-Reply-To: <b0a271780901201043te516f6dm5e3699e7e4d160c4@mail.gmail.com>
References: <b0a271780901201043te516f6dm5e3699e7e4d160c4@mail.gmail.com>
Message-ID: <191c3a030901201054t39b977aar39859282fc72aa28@mail.gmail.com>

is it your own custom event?

switch_channel_event_set_data(channel, event);

in your code will add all the vars.


On Tue, Jan 20, 2009 at 12:43 PM, Johny Kadarisman <jkr888 at gmail.com> wrote:

> Hi all,
>
> Is there a way to propagate variable_* into custom event. I'm
> connecting with inbound socket, and listening to that custom events.
>
> during recent trunk update, I encountered an issues from external
> application, after look around, found out that these value is not in
> the events. After search around, found out some reference about
> "verbose_events" with outbound sockets. So, I insert that applications
> in dialplan, but still don't see any variable_* being propagate.
>
> Is there anyway to enable this for inbound sockets?
>
> Thanks,
> Johny K.
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jkr888 at gmail.com  Tue Jan 20 11:07:35 2009
From: jkr888 at gmail.com (Johny Kadarisman)
Date: Tue, 20 Jan 2009 14:07:35 -0500
Subject: [Freeswitch-dev] Latest build - Socket inbound variable_* on
	custom event problem.
In-Reply-To: <191c3a030901201054t39b977aar39859282fc72aa28@mail.gmail.com>
References: <b0a271780901201043te516f6dm5e3699e7e4d160c4@mail.gmail.com>
	<191c3a030901201054t39b977aar39859282fc72aa28@mail.gmail.com>
Message-ID: <b0a271780901201107q4ae5be1fya719e60c097933e0@mail.gmail.com>

The external socket apps listen on the custom event that generate by
mod_conference.
It uses some customize variable_* that being passes in the event to do
certain logic.

Can i enable the propagation externally from dialplan, similar to
"verbose_events" for outbound?

Johny K.

On Tue, Jan 20, 2009 at 1:54 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> is it your own custom event?
>
> switch_channel_event_set_data(channel, event);
>
> in your code will add all the vars.
>
>
> On Tue, Jan 20, 2009 at 12:43 PM, Johny Kadarisman <jkr888 at gmail.com> wrote:
>>
>> Hi all,
>>
>> Is there a way to propagate variable_* into custom event. I'm
>> connecting with inbound socket, and listening to that custom events.
>>
>> during recent trunk update, I encountered an issues from external
>> application, after look around, found out that these value is not in
>> the events. After search around, found out some reference about
>> "verbose_events" with outbound sockets. So, I insert that applications
>> in dialplan, but still don't see any variable_* being propagate.
>>
>> Is there anyway to enable this for inbound sockets?
>>
>> Thanks,
>> Johny K.
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From jkr888 at gmail.com  Tue Jan 20 11:18:22 2009
From: jkr888 at gmail.com (Johny Kadarisman)
Date: Tue, 20 Jan 2009 14:18:22 -0500
Subject: [Freeswitch-dev] Latest build - Socket inbound variable_* on
	custom event problem.
In-Reply-To: <b0a271780901201107q4ae5be1fya719e60c097933e0@mail.gmail.com>
References: <b0a271780901201043te516f6dm5e3699e7e4d160c4@mail.gmail.com>
	<191c3a030901201054t39b977aar39859282fc72aa28@mail.gmail.com>
	<b0a271780901201107q4ae5be1fya719e60c097933e0@mail.gmail.com>
Message-ID: <b0a271780901201118n48240611o75aeb4779ef0961@mail.gmail.com>

I think i found it, it can be configure from conference profile :)

Thanks for the pointer.

On Tue, Jan 20, 2009 at 2:07 PM, Johny Kadarisman <jkr888 at gmail.com> wrote:
> The external socket apps listen on the custom event that generate by
> mod_conference.
> It uses some customize variable_* that being passes in the event to do
> certain logic.
>
> Can i enable the propagation externally from dialplan, similar to
> "verbose_events" for outbound?
>
> Johny K.
>
> On Tue, Jan 20, 2009 at 1:54 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> is it your own custom event?
>>
>> switch_channel_event_set_data(channel, event);
>>
>> in your code will add all the vars.
>>
>>
>> On Tue, Jan 20, 2009 at 12:43 PM, Johny Kadarisman <jkr888 at gmail.com> wrote:
>>>
>>> Hi all,
>>>
>>> Is there a way to propagate variable_* into custom event. I'm
>>> connecting with inbound socket, and listening to that custom events.
>>>
>>> during recent trunk update, I encountered an issues from external
>>> application, after look around, found out that these value is not in
>>> the events. After search around, found out some reference about
>>> "verbose_events" with outbound sockets. So, I insert that applications
>>> in dialplan, but still don't see any variable_* being propagate.
>>>
>>> Is there anyway to enable this for inbound sockets?
>>>
>>> Thanks,
>>> Johny K.
>>>
>>> _______________________________________________
>>> Freeswitch-dev mailing list
>>> Freeswitch-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>


From jkr888 at gmail.com  Tue Jan 20 11:29:55 2009
From: jkr888 at gmail.com (Johny Kadarisman)
Date: Tue, 20 Jan 2009 14:29:55 -0500
Subject: [Freeswitch-dev] Latest build - Socket inbound variable_* on
	custom event problem.
In-Reply-To: <b0a271780901201118n48240611o75aeb4779ef0961@mail.gmail.com>
References: <b0a271780901201043te516f6dm5e3699e7e4d160c4@mail.gmail.com>
	<191c3a030901201054t39b977aar39859282fc72aa28@mail.gmail.com>
	<b0a271780901201107q4ae5be1fya719e60c097933e0@mail.gmail.com>
	<b0a271780901201118n48240611o75aeb4779ef0961@mail.gmail.com>
Message-ID: <b0a271780901201129r6b9a39eds32f70381372d011b@mail.gmail.com>

just realize that this settings is not in the default conference
configuration, so if someone have the same issues like me, you can
save your debugging time and try put following line in conference
configuration ;)

      <param name="verbose-events" value="true"/>


On Tue, Jan 20, 2009 at 2:18 PM, Johny Kadarisman <jkr888 at gmail.com> wrote:
> I think i found it, it can be configure from conference profile :)
>
> Thanks for the pointer.
>
> On Tue, Jan 20, 2009 at 2:07 PM, Johny Kadarisman <jkr888 at gmail.com> wrote:
>> The external socket apps listen on the custom event that generate by
>> mod_conference.
>> It uses some customize variable_* that being passes in the event to do
>> certain logic.
>>
>> Can i enable the propagation externally from dialplan, similar to
>> "verbose_events" for outbound?
>>
>> Johny K.
>>
>> On Tue, Jan 20, 2009 at 1:54 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> is it your own custom event?
>>>
>>> switch_channel_event_set_data(channel, event);
>>>
>>> in your code will add all the vars.
>>>
>>>
>>> On Tue, Jan 20, 2009 at 12:43 PM, Johny Kadarisman <jkr888 at gmail.com> wrote:
>>>>
>>>> Hi all,
>>>>
>>>> Is there a way to propagate variable_* into custom event. I'm
>>>> connecting with inbound socket, and listening to that custom events.
>>>>
>>>> during recent trunk update, I encountered an issues from external
>>>> application, after look around, found out that these value is not in
>>>> the events. After search around, found out some reference about
>>>> "verbose_events" with outbound sockets. So, I insert that applications
>>>> in dialplan, but still don't see any variable_* being propagate.
>>>>
>>>> Is there anyway to enable this for inbound sockets?
>>>>
>>>> Thanks,
>>>> Johny K.
>>>>
>>>> _______________________________________________
>>>> Freeswitch-dev mailing list
>>>> Freeswitch-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> Freeswitch-dev mailing list
>>> Freeswitch-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>>
>>
>


From seven at idapted.com  Tue Jan 20 17:58:07 2009
From: seven at idapted.com (seven du)
Date: Wed, 21 Jan 2009 09:58:07 +0800
Subject: [Freeswitch-dev] Is there any plans to implement some kind of
	measures and statistics function?
In-Reply-To: <4975A190.8080001@gmail.com>
References: <BD056EFB-86AE-4E6C-8020-7197511FE6AA@idapted.com>
	<4975A190.8080001@gmail.com>
Message-ID: <C21418F8-6262-43AC-BA90-C7B8C54D50C3@idapted.com>

Hi Tamas,

Thank you for responding, I'm not sure, but if I can add some $$$, how  
much do you think it will be enough?

As you know, about the RTCP protocol, even if if can be implemented,  
it must be supported by both ends.
Actually what I currently think is:  can we just let FS report this  
kind of message(via event_socket of CDR)?
Is it enough for you?

I want to read the FS code and see how easy/hard to do this. It will  
be really helpful if someone can give some hints to do that.

Best,
   Seven


On Jan 20, 2009, at 6:04 PM, Tamas wrote:

> Hello Seven,
>
> We are the one put the bounty :)
> Do you mind to add some $$$ to the fund? That could help the thing  
> make
> happen.
>
> Regards,
>    Tamas
>
> ps: Right waiting for response from consulting@ for the request.
>
> seven du ?rta:
>> Hi developers,
>>
>> Now and then we need to know some statistics message and sure we need
>> some way to measure it. My ideas just like this:
>>
>> http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Extended_Reports_.28RTCP_XR.29_support
>>
>> Is there some plans to build this kind of function in? If not, can
>> some one tell me how easy or hard to do it? I think it would be  
>> better
>> to use pcap or 3rd party stuffs the do that.
>>
>> Thanks.
>>
>>
>> Seven
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> dev
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From msc at freeswitch.org  Tue Jan 20 21:11:43 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Jan 2009 21:11:43 -0800
Subject: [Freeswitch-dev] New wiki page needs your attention!
Message-ID: <87f2f3b90901202111x3811ba36ve2ec3ecda31c6855@mail.gmail.com>

Hello FreeSWITCHers!

I'm putting the finishing touches on a wiki page that we hope will
make it easier for users to request help and for the dev team and
power users to digest and process those requests. What I need first
and foremost is for everyone to please read this page:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

Please give me feedback. Put yourself in the shoes of a relative
newbie. Is the information easy to follow? On the flip side, if you
wanted to help someone, ask yourself, if they follow the steps on this
page will that suffice? Are there places that need improvement? Can
you think of anything else that can be added?

NOTE: I'm still working on the TDM/OpenZAP section as well as the
sections on scripting, event socket, elements of a jira ticket, etc.
If you have suggestions for content on those pages please email me off
list or hop and and fill in some of the blanks.

The core development team really appreciates all of your help. Now
that FS is growing like mad we are at the point where it is imperative
that we have reliable documentation for new ones so that the
developers and other experts can focus on advancing the project even
further. Let's all lend a hand by improving the documentation.

Thanks again!
-MC (mercutioviz)


From sanju at 11hit.com  Wed Jan 21 02:16:22 2009
From: sanju at 11hit.com (Sanju)
Date: Wed, 21 Jan 2009 02:16:22 -0800
Subject: [Freeswitch-dev] Reg: Freeswitch execution error [urgent]
Message-ID: <20090121021622.F6C0A77D@resin13.mta.everyone.net>

Hi,
  I have Installed Freeswitch in Redhat5.1 , but when i start the freeswitch for /usr/local/freeswitch/bin throws the following error

2009-01-21 13:54:01 [ERR] sofia_glue.c:559 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Timeout]
2009-01-21 13:54:01 [ERR] sofia.c:1815 config_sofia() Failed to get external ip.

i am able to ping stun.freeswitch.org but not allowing when specified stun.freeswitch.org:3478 , the linux doesn't have any firewall and connected with internet 

please help me , do i need to configure any thing else ?


Regards, 
Sanju. 



_____________________________________________________________
Gift Certificates
http://www.online-gift-certificate.com

This email was sent using 11hit.com free web-based email!
http://www.11hit.com


From rob.charlton at savageminds.com  Wed Jan 21 04:38:54 2009
From: rob.charlton at savageminds.com (Rob Charlton)
Date: Wed, 21 Jan 2009 12:38:54 +0000
Subject: [Freeswitch-dev] DTMF events
Message-ID: <4977175E.1020303@savageminds.com>

Hi,

I'm using mod_event_socket to listen for DTMF events. I have Nokia 
handsets registered as SIP clients over Wifi, as well as a SIP trunk 
providing incoming PSTN calls to a range of DDIs and outgoing PSTN calls.

If I make an incoming (PSTN or SIP) call and answer it, I always see 
DTMF events via mod_event_socket.
If I make an outgoing call direct to a handset using SIP then I see DTMF 
events - e.g. originate user/1000 &park()
If I make an outgoing call via PSTN then I don't see DTMF events e.g. 
originate sofia/gateway/mygateway/myphonenumber &park() or 
&javascript(myscript.js);

In the latter case, I am still able to pick up DTMF digits if I use 
javascript session.collectInput() - so it appears as if the DTMF tones 
are being recognised by Freeswitch, but no events sent.

What am I doing wrong?

Cheers

Rob

-- 
Rob Charlton
Savage Minds Ltd




From anthony.minessale at gmail.com  Wed Jan 21 05:50:57 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Jan 2009 07:50:57 -0600
Subject: [Freeswitch-dev] Reg: Freeswitch execution error [urgent]
In-Reply-To: <20090121021622.F6C0A77D@resin13.mta.everyone.net>
References: <20090121021622.F6C0A77D@resin13.mta.everyone.net>
Message-ID: <191c3a030901210550l5cced4bct2b77798843197028@mail.gmail.com>

this isn't really a dev topic this is more appropriate for freeswitch-users
you just have to delete all the profiles besides internal


On Wed, Jan 21, 2009 at 4:16 AM, Sanju <sanju at 11hit.com> wrote:

> Hi,
>  I have Installed Freeswitch in Redhat5.1 , but when i start the freeswitch
> for /usr/local/freeswitch/bin throws the following error
>
> 2009-01-21 13:54:01 [ERR] sofia_glue.c:559 sofia_glue_ext_address_lookup()
> STUN Failed! stun.freeswitch.org:3478 [Timeout]
> 2009-01-21 13:54:01 [ERR] sofia.c:1815 config_sofia() Failed to get
> external ip.
>
> i am able to ping stun.freeswitch.org but not allowing when specified
> stun.freeswitch.org:3478 , the linux doesn't have any firewall and
> connected with internet
>
> please help me , do i need to configure any thing else ?
>
>
> Regards,
> Sanju.
>
>
>
> _____________________________________________________________
> Gift Certificates
> http://www.online-gift-certificate.com
>
> This email was sent using 11hit.com free web-based email!
> http://www.11hit.com
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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From anthony.minessale at gmail.com  Wed Jan 21 06:00:32 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Jan 2009 08:00:32 -0600
Subject: [Freeswitch-dev] DTMF events
In-Reply-To: <4977175E.1020303@savageminds.com>
References: <4977175E.1020303@savageminds.com>
Message-ID: <191c3a030901210600m7437e4e7n91d804c01720b3e9@mail.gmail.com>

Did you try enabling all events and making a single call to make sure you
are subscribed to the right event?


On Wed, Jan 21, 2009 at 6:38 AM, Rob Charlton
<rob.charlton at savageminds.com>wrote:

> Hi,
>
> I'm using mod_event_socket to listen for DTMF events. I have Nokia
> handsets registered as SIP clients over Wifi, as well as a SIP trunk
> providing incoming PSTN calls to a range of DDIs and outgoing PSTN calls.
>
> If I make an incoming (PSTN or SIP) call and answer it, I always see
> DTMF events via mod_event_socket.
> If I make an outgoing call direct to a handset using SIP then I see DTMF
> events - e.g. originate user/1000 &park()
> If I make an outgoing call via PSTN then I don't see DTMF events e.g.
> originate sofia/gateway/mygateway/myphonenumber &park() or
> &javascript(myscript.js);
>
> In the latter case, I am still able to pick up DTMF digits if I use
> javascript session.collectInput() - so it appears as if the DTMF tones
> are being recognised by Freeswitch, but no events sent.
>
> What am I doing wrong?
>
> Cheers
>
> Rob
>
> --
> Rob Charlton
> Savage Minds Ltd
>
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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From dujinfang at gmail.com  Tue Jan 20 22:50:36 2009
From: dujinfang at gmail.com (dujinfang)
Date: Wed, 21 Jan 2009 14:50:36 +0800
Subject: [Freeswitch-dev] New wiki page needs your attention!
In-Reply-To: <87f2f3b90901202111x3811ba36ve2ec3ecda31c6855@mail.gmail.com>
References: <87f2f3b90901202111x3811ba36ve2ec3ecda31c6855@mail.gmail.com>
Message-ID: <8E79CA7D-5CF3-4C04-8DE1-F464E9E9BEA6@gmail.com>

Great!  but how can I login into pastebin?

On Jan 21, 2009, at 1:11 PM, Michael Collins wrote:

> Hello FreeSWITCHers!
>
> I'm putting the finishing touches on a wiki page that we hope will
> make it easier for users to request help and for the dev team and
> power users to digest and process those requests. What I need first
> and foremost is for everyone to please read this page:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
> Please give me feedback. Put yourself in the shoes of a relative
> newbie. Is the information easy to follow? On the flip side, if you
> wanted to help someone, ask yourself, if they follow the steps on this
> page will that suffice? Are there places that need improvement? Can
> you think of anything else that can be added?
>
> NOTE: I'm still working on the TDM/OpenZAP section as well as the
> sections on scripting, event socket, elements of a jira ticket, etc.
> If you have suggestions for content on those pages please email me off
> list or hop and and fill in some of the blanks.
>
> The core development team really appreciates all of your help. Now
> that FS is growing like mad we are at the point where it is imperative
> that we have reliable documentation for new ones so that the
> developers and other experts can focus on advancing the project even
> further. Let's all lend a hand by improving the documentation.
>
> Thanks again!
> -MC (mercutioviz)
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From msc at freeswitch.org  Wed Jan 21 06:39:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Jan 2009 06:39:29 -0800
Subject: [Freeswitch-dev] New wiki page needs your attention!
In-Reply-To: <8E79CA7D-5CF3-4C04-8DE1-F464E9E9BEA6@gmail.com>
References: <87f2f3b90901202111x3811ba36ve2ec3ecda31c6855@mail.gmail.com>
	<8E79CA7D-5CF3-4C04-8DE1-F464E9E9BEA6@gmail.com>
Message-ID: <87f2f3b90901210639i76555732t4f688f689cc890d4@mail.gmail.com>

On Tue, Jan 20, 2009 at 10:50 PM, dujinfang <dujinfang at gmail.com> wrote:
> Great!  but how can I login into pastebin?

Look closely at the login dialog box and you'll see the answer to your
question! ;)
-MC

>
> On Jan 21, 2009, at 1:11 PM, Michael Collins wrote:
>
>> Hello FreeSWITCHers!
>>
>> I'm putting the finishing touches on a wiki page that we hope will
>> make it easier for users to request help and for the dev team and
>> power users to digest and process those requests. What I need first
>> and foremost is for everyone to please read this page:
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>
>> Please give me feedback. Put yourself in the shoes of a relative
>> newbie. Is the information easy to follow? On the flip side, if you
>> wanted to help someone, ask yourself, if they follow the steps on this
>> page will that suffice? Are there places that need improvement? Can
>> you think of anything else that can be added?
>>
>> NOTE: I'm still working on the TDM/OpenZAP section as well as the
>> sections on scripting, event socket, elements of a jira ticket, etc.
>> If you have suggestions for content on those pages please email me off
>> list or hop and and fill in some of the blanks.
>>
>> The core development team really appreciates all of your help. Now
>> that FS is growing like mad we are at the point where it is imperative
>> that we have reliable documentation for new ones so that the
>> developers and other experts can focus on advancing the project even
>> further. Let's all lend a hand by improving the documentation.
>>
>> Thanks again!
>> -MC (mercutioviz)
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From msc at freeswitch.org  Wed Jan 21 06:47:39 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Jan 2009 06:47:39 -0800
Subject: [Freeswitch-dev] DTMF events
In-Reply-To: <191c3a030901210600m7437e4e7n91d804c01720b3e9@mail.gmail.com>
References: <4977175E.1020303@savageminds.com>
	<191c3a030901210600m7437e4e7n91d804c01720b3e9@mail.gmail.com>
Message-ID: <87f2f3b90901210647m657d90c1me408fcd5ea394230@mail.gmail.com>

FYI,
I've added a bit to the reporting bugs page to discuss this. Still
needs more, but it's getting there.
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Event_Socket

-MC

On Wed, Jan 21, 2009 at 6:00 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> Did you try enabling all events and making a single call to make sure you
> are subscribed to the right event?
>
>
> On Wed, Jan 21, 2009 at 6:38 AM, Rob Charlton <rob.charlton at savageminds.com>
> wrote:
>>
>> Hi,
>>
>> I'm using mod_event_socket to listen for DTMF events. I have Nokia
>> handsets registered as SIP clients over Wifi, as well as a SIP trunk
>> providing incoming PSTN calls to a range of DDIs and outgoing PSTN calls.
>>
>> If I make an incoming (PSTN or SIP) call and answer it, I always see
>> DTMF events via mod_event_socket.
>> If I make an outgoing call direct to a handset using SIP then I see DTMF
>> events - e.g. originate user/1000 &park()
>> If I make an outgoing call via PSTN then I don't see DTMF events e.g.
>> originate sofia/gateway/mygateway/myphonenumber &park() or
>> &javascript(myscript.js);
>>
>> In the latter case, I am still able to pick up DTMF digits if I use
>> javascript session.collectInput() - so it appears as if the DTMF tones
>> are being recognised by Freeswitch, but no events sent.
>>
>> What am I doing wrong?
>>
>> Cheers
>>
>> Rob
>>
>> --
>> Rob Charlton
>> Savage Minds Ltd
>>
>>
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From rob.charlton at savageminds.com  Wed Jan 21 07:53:41 2009
From: rob.charlton at savageminds.com (Rob Charlton)
Date: Wed, 21 Jan 2009 15:53:41 +0000
Subject: [Freeswitch-dev] DTMF events
In-Reply-To: <191c3a030901210600m7437e4e7n91d804c01720b3e9@mail.gmail.com>
References: <4977175E.1020303@savageminds.com>
	<191c3a030901210600m7437e4e7n91d804c01720b3e9@mail.gmail.com>
Message-ID: <49774505.3050400@savageminds.com>

Yes, and yes. I see the DTMF events arriving when I make an incoming call.

I put a breakpoint on switch_channel_dequeue_dtmf() which gets hit when 
I type digits after:

- I originate a call to a sip extension
- I receive a call from a sip extension
- I receive a call from our sip trunk (from PSTN)

The breakpoint doesn't get hit when I type digits after:

- I originate a call via our sip trunk (to the PSTN)

As regards this:
 > In the latter case, I am still able to pick up DTMF digits if I use
 > javascript session.collectInput() - so it appears as if the DTMF tones
 > are being recognised by Freeswitch, but no events sent.
I must have been dreaming - that isn't the case at all - 
session.collectInput doesn't get any digits at all.

We use the same SIP trunk with asterisk and that _does_ pick up DTMF 
tones for outbound PSTN calls.

Thanks

Rob


Anthony Minessale wrote:
> Did you try enabling all events and making a single call to make sure 
> you are subscribed to the right event?
>
>
> On Wed, Jan 21, 2009 at 6:38 AM, Rob Charlton 
> <rob.charlton at savageminds.com <mailto:rob.charlton at savageminds.com>> 
> wrote:
>
>     Hi,
>
>     I'm using mod_event_socket to listen for DTMF events. I have Nokia
>     handsets registered as SIP clients over Wifi, as well as a SIP trunk
>     providing incoming PSTN calls to a range of DDIs and outgoing PSTN
>     calls.
>
>     If I make an incoming (PSTN or SIP) call and answer it, I always see
>     DTMF events via mod_event_socket.
>     If I make an outgoing call direct to a handset using SIP then I
>     see DTMF
>     events - e.g. originate user/1000 &park()
>     If I make an outgoing call via PSTN then I don't see DTMF events e.g.
>     originate sofia/gateway/mygateway/myphonenumber &park() or
>     &javascript(myscript.js);
>
>     In the latter case, I am still able to pick up DTMF digits if I use
>     javascript session.collectInput() - so it appears as if the DTMF tones
>     are being recognised by Freeswitch, but no events sent.
>
>     What am I doing wrong?
>
>     Cheers
>
>     Rob
>
>     --
>     Rob Charlton
>     Savage Minds Ltd
>



From ludovic.fouquet at bewan.com  Wed Jan 21 09:44:02 2009
From: ludovic.fouquet at bewan.com (ludovic)
Date: Wed, 21 Jan 2009 18:44:02 +0100
Subject: [Freeswitch-dev] switch_frame structure question
Message-ID: <49775EE2.1060203@bewan.com>

Hi,

what is the difference in the structure switch_frame between the 
following members :
packet
data
payload

The routine  channel_write_frame of my endpoind is  called, but it seems 
that I do not use the correct data as my payload_type of the data is not 
8 (PCMA) which I should have.
I call from a SIP phone, registered in freeswitch to an analog phone.
I have to pass the RTP packet to my phone device.

Thanks,

Ludovic




From anthony.minessale at gmail.com  Wed Jan 21 09:54:30 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Jan 2009 11:54:30 -0600
Subject: [Freeswitch-dev] switch_frame structure question
In-Reply-To: <49775EE2.1060203@bewan.com>
References: <49775EE2.1060203@bewan.com>
Message-ID: <191c3a030901210954y41acc7c7r952d5021ddb5a5c8@mail.gmail.com>

packet is the raw packet pointer
if the frame flag SFF_RAW_RTP is set it can be cast into a rtp header to
view the contents.
data is the media (audio data)
payload is sometimes used to describe the payload type of the packet.

if you are looking to see what the audio type is

try frame->codec->implementation->ianacode and
frame->codec->implementation->iananame


On Wed, Jan 21, 2009 at 11:44 AM, ludovic <ludovic.fouquet at bewan.com> wrote:

> Hi,
>
> what is the difference in the structure switch_frame between the
> following members :
> packet
> data
> payload
>
> The routine  channel_write_frame of my endpoind is  called, but it seems
> that I do not use the correct data as my payload_type of the data is not
> 8 (PCMA) which I should have.
> I call from a SIP phone, registered in freeswitch to an analog phone.
> I have to pass the RTP packet to my phone device.
>
> Thanks,
>
> Ludovic
>
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Wed Jan 21 13:19:05 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Jan 2009 13:19:05 -0800
Subject: [Freeswitch-dev] DTMF events
In-Reply-To: <49774505.3050400@savageminds.com>
References: <4977175E.1020303@savageminds.com>
	<191c3a030901210600m7437e4e7n91d804c01720b3e9@mail.gmail.com>
	<49774505.3050400@savageminds.com>
Message-ID: <87f2f3b90901211319p68f2d28cr178fc357cd6dcd77@mail.gmail.com>

Rob,

I've been able to duplicate this behavior on my Mac with r11333. It
seems to work with Lua but not with Javascript. I am going to discuss
it with the devs and possibly open a jira issue. In the meantime would
you be willing to try it with Lua, even just for testing? This worked
for me:
<lua script>
-- Test sending custom events in Lua
local event = freeswitch.Event("custom");
event:addHeader("Sample Custom Event", "no");
event:fire();
</lua script>
I saved the above as /usr/local/freeswitch/scripts/event1.lua

I then opened two terminal windows, one to freeswitch CLI and the
other a telnet into the event socket
On the event socket I logged in and listened for custom events:
 telnet localhost 8021 <enter>
 auth ClueCon <enter><enter>
 events plain custom <enter><enter>

On FS CLI I typed:
 lua event1.lua <press enter>

On the event socket I immediately see this:
Sample Custom Event: no
Event-Name: CUSTOM
Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc
FreeSWITCH-Hostname: michael-collinss-macbook-pro.local
FreeSWITCH-IPv4: 192.168.1.5
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-01-21%2013%3A14%3A35
Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A14%3A35%20GMT
Event-Date-Timestamp: 1232572475813346
Event-Calling-File: switch_cpp.cpp
Event-Calling-Function: fire
Event-Calling-Line-Number: 295

However, when I do the same kind of thing with js it doesn't work:
<js script>
// Sample event sent from JavaScript
console_log("INFO","Starting event1.js sample event sender...\n");
    var msg = "Hello, welcome to the FreeSWITCH demo application
please enter some text into the chat box";
    e = new Event("custom", "message");
    e.addBody(msg);
    e.fire();
</js script>

I saved the above as /usr/local/freeswitch/scripts/event1.js
I run it from the FS CLI:
 jsrun event1.js <enter>
And I see my console message pop up but I don't see anything on the event socket
However, if I do this at the event socket:
 events plain all <enter><enter>

And then do jsrun event1.js from FS CLI then I do see my event on the
event socket like this:
Content-Length: 559
Content-Type: text/event-plain

Event-Subclass: message
Event-Name: CUSTOM
Core-UUID: 2c04a36e-5a23-4b14-b0a5-34fe9fd9f1bc
FreeSWITCH-Hostname: michael-collinss-macbook-pro.local
FreeSWITCH-IPv4: 192.168.1.5
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-01-21%2013%3A07%3A48
Event-Date-GMT: Wed,%2021%20Jan%202009%2021%3A07%3A48%20GMT
Event-Date-Timestamp: 1232572068370864
Event-Calling-File: mod_spidermonkey.c
Event-Calling-Function: event_fire
Event-Calling-Line-Number: 671
Content-Length: 90

Hello, welcome to the FreeSWITCH demo application please enter some
text into the chat box

So, there's definitely something going on, we just need to find out
what for sure. I'll be in touch.
-MC (mercutioviz)
On Wed, Jan 21, 2009 at 7:53 AM, Rob Charlton
<rob.charlton at savageminds.com> wrote:
> Yes, and yes. I see the DTMF events arriving when I make an incoming call.
>
> I put a breakpoint on switch_channel_dequeue_dtmf() which gets hit when
> I type digits after:
>
> - I originate a call to a sip extension
> - I receive a call from a sip extension
> - I receive a call from our sip trunk (from PSTN)
>
> The breakpoint doesn't get hit when I type digits after:
>
> - I originate a call via our sip trunk (to the PSTN)
>
> As regards this:
>  > In the latter case, I am still able to pick up DTMF digits if I use
>  > javascript session.collectInput() - so it appears as if the DTMF tones
>  > are being recognised by Freeswitch, but no events sent.
> I must have been dreaming - that isn't the case at all -
> session.collectInput doesn't get any digits at all.
>
> We use the same SIP trunk with asterisk and that _does_ pick up DTMF
> tones for outbound PSTN calls.
>
> Thanks
>
> Rob
>
>
> Anthony Minessale wrote:
>> Did you try enabling all events and making a single call to make sure
>> you are subscribed to the right event?
>>
>>
>> On Wed, Jan 21, 2009 at 6:38 AM, Rob Charlton
>> <rob.charlton at savageminds.com <mailto:rob.charlton at savageminds.com>>
>> wrote:
>>
>>     Hi,
>>
>>     I'm using mod_event_socket to listen for DTMF events. I have Nokia
>>     handsets registered as SIP clients over Wifi, as well as a SIP trunk
>>     providing incoming PSTN calls to a range of DDIs and outgoing PSTN
>>     calls.
>>
>>     If I make an incoming (PSTN or SIP) call and answer it, I always see
>>     DTMF events via mod_event_socket.
>>     If I make an outgoing call direct to a handset using SIP then I
>>     see DTMF
>>     events - e.g. originate user/1000 &park()
>>     If I make an outgoing call via PSTN then I don't see DTMF events e.g.
>>     originate sofia/gateway/mygateway/myphonenumber &park() or
>>     &javascript(myscript.js);
>>
>>     In the latter case, I am still able to pick up DTMF digits if I use
>>     javascript session.collectInput() - so it appears as if the DTMF tones
>>     are being recognised by Freeswitch, but no events sent.
>>
>>     What am I doing wrong?
>>
>>     Cheers
>>
>>     Rob
>>
>>     --
>>     Rob Charlton
>>     Savage Minds Ltd
>>
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From msc at freeswitch.org  Wed Jan 21 13:41:14 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Jan 2009 13:41:14 -0800
Subject: [Freeswitch-dev] DTMF events
In-Reply-To: <49774505.3050400@savageminds.com>
References: <4977175E.1020303@savageminds.com>
	<191c3a030901210600m7437e4e7n91d804c01720b3e9@mail.gmail.com>
	<49774505.3050400@savageminds.com>
Message-ID: <87f2f3b90901211341w368d9437q6e09374a55c824cf@mail.gmail.com>

Rob,
My bad, I replied to the wrong email thread. Please disregard.
-MC

On Wed, Jan 21, 2009 at 7:53 AM, Rob Charlton
<rob.charlton at savageminds.com> wrote:
> Yes, and yes. I see the DTMF events arriving when I make an incoming call.
>
> I put a breakpoint on switch_channel_dequeue_dtmf() which gets hit when
> I type digits after:
>
> - I originate a call to a sip extension
> - I receive a call from a sip extension
> - I receive a call from our sip trunk (from PSTN)
>
> The breakpoint doesn't get hit when I type digits after:
>
> - I originate a call via our sip trunk (to the PSTN)
>
> As regards this:
>  > In the latter case, I am still able to pick up DTMF digits if I use
>  > javascript session.collectInput() - so it appears as if the DTMF tones
>  > are being recognised by Freeswitch, but no events sent.
> I must have been dreaming - that isn't the case at all -
> session.collectInput doesn't get any digits at all.
>
> We use the same SIP trunk with asterisk and that _does_ pick up DTMF
> tones for outbound PSTN calls.
>
> Thanks
>
> Rob
>
>
> Anthony Minessale wrote:
>> Did you try enabling all events and making a single call to make sure
>> you are subscribed to the right event?
>>
>>
>> On Wed, Jan 21, 2009 at 6:38 AM, Rob Charlton
>> <rob.charlton at savageminds.com <mailto:rob.charlton at savageminds.com>>
>> wrote:
>>
>>     Hi,
>>
>>     I'm using mod_event_socket to listen for DTMF events. I have Nokia
>>     handsets registered as SIP clients over Wifi, as well as a SIP trunk
>>     providing incoming PSTN calls to a range of DDIs and outgoing PSTN
>>     calls.
>>
>>     If I make an incoming (PSTN or SIP) call and answer it, I always see
>>     DTMF events via mod_event_socket.
>>     If I make an outgoing call direct to a handset using SIP then I
>>     see DTMF
>>     events - e.g. originate user/1000 &park()
>>     If I make an outgoing call via PSTN then I don't see DTMF events e.g.
>>     originate sofia/gateway/mygateway/myphonenumber &park() or
>>     &javascript(myscript.js);
>>
>>     In the latter case, I am still able to pick up DTMF digits if I use
>>     javascript session.collectInput() - so it appears as if the DTMF tones
>>     are being recognised by Freeswitch, but no events sent.
>>
>>     What am I doing wrong?
>>
>>     Cheers
>>
>>     Rob
>>
>>     --
>>     Rob Charlton
>>     Savage Minds Ltd
>>
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Wed Jan 21 13:43:23 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Jan 2009 15:43:23 -0600
Subject: [Freeswitch-dev] DTMF events
In-Reply-To: <49774505.3050400@savageminds.com>
References: <4977175E.1020303@savageminds.com>
	<191c3a030901210600m7437e4e7n91d804c01720b3e9@mail.gmail.com>
	<49774505.3050400@savageminds.com>
Message-ID: <191c3a030901211343x4c616c2emcdde76f91c035f33@mail.gmail.com>

when you say tones, does that mean it's inband dtmf?
you may need to run the start_dtmf app on the channel to engage the tone
detector?

can you please file it on jira http://jira.freeswitch.org and attach
a pcap and console debug log.



On Wed, Jan 21, 2009 at 9:53 AM, Rob Charlton
<rob.charlton at savageminds.com>wrote:

> Yes, and yes. I see the DTMF events arriving when I make an incoming call.
>
> I put a breakpoint on switch_channel_dequeue_dtmf() which gets hit when
> I type digits after:
>
> - I originate a call to a sip extension
> - I receive a call from a sip extension
> - I receive a call from our sip trunk (from PSTN)
>
> The breakpoint doesn't get hit when I type digits after:
>
> - I originate a call via our sip trunk (to the PSTN)
>
> As regards this:
>  > In the latter case, I am still able to pick up DTMF digits if I use
>  > javascript session.collectInput() - so it appears as if the DTMF tones
>  > are being recognised by Freeswitch, but no events sent.
> I must have been dreaming - that isn't the case at all -
> session.collectInput doesn't get any digits at all.
>
> We use the same SIP trunk with asterisk and that _does_ pick up DTMF
> tones for outbound PSTN calls.
>
> Thanks
>
> Rob
>
>
> Anthony Minessale wrote:
> > Did you try enabling all events and making a single call to make sure
> > you are subscribed to the right event?
> >
> >
> > On Wed, Jan 21, 2009 at 6:38 AM, Rob Charlton
> > <rob.charlton at savageminds.com <mailto:rob.charlton at savageminds.com>>
> > wrote:
> >
> >     Hi,
> >
> >     I'm using mod_event_socket to listen for DTMF events. I have Nokia
> >     handsets registered as SIP clients over Wifi, as well as a SIP trunk
> >     providing incoming PSTN calls to a range of DDIs and outgoing PSTN
> >     calls.
> >
> >     If I make an incoming (PSTN or SIP) call and answer it, I always see
> >     DTMF events via mod_event_socket.
> >     If I make an outgoing call direct to a handset using SIP then I
> >     see DTMF
> >     events - e.g. originate user/1000 &park()
> >     If I make an outgoing call via PSTN then I don't see DTMF events e.g.
> >     originate sofia/gateway/mygateway/myphonenumber &park() or
> >     &javascript(myscript.js);
> >
> >     In the latter case, I am still able to pick up DTMF digits if I use
> >     javascript session.collectInput() - so it appears as if the DTMF
> tones
> >     are being recognised by Freeswitch, but no events sent.
> >
> >     What am I doing wrong?
> >
> >     Cheers
> >
> >     Rob
> >
> >     --
> >     Rob Charlton
> >     Savage Minds Ltd
> >
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From dujinfang at gmail.com  Wed Jan 21 15:15:11 2009
From: dujinfang at gmail.com (dujinfang)
Date: Thu, 22 Jan 2009 07:15:11 +0800
Subject: [Freeswitch-dev] New wiki page needs your attention!
In-Reply-To: <87f2f3b90901210639i76555732t4f688f689cc890d4@mail.gmail.com>
References: <87f2f3b90901202111x3811ba36ve2ec3ecda31c6855@mail.gmail.com>
	<8E79CA7D-5CF3-4C04-8DE1-F464E9E9BEA6@gmail.com>
	<87f2f3b90901210639i76555732t4f688f689cc890d4@mail.gmail.com>
Message-ID: <F4A64BF7-45FB-48BC-B203-4CE39B0FA354@gmail.com>

Thanks, silly question though :)

On Jan 21, 2009, at 10:39 PM, Michael Collins wrote:

> On Tue, Jan 20, 2009 at 10:50 PM, dujinfang <dujinfang at gmail.com>  
> wrote:
>> Great!  but how can I login into pastebin?
>
> Look closely at the login dialog box and you'll see the answer to your
> question! ;)
> -MC
>
>>
>> On Jan 21, 2009, at 1:11 PM, Michael Collins wrote:
>>
>>> Hello FreeSWITCHers!
>>>
>>> I'm putting the finishing touches on a wiki page that we hope will
>>> make it easier for users to request help and for the dev team and
>>> power users to digest and process those requests. What I need first
>>> and foremost is for everyone to please read this page:
>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>
>>> Please give me feedback. Put yourself in the shoes of a relative
>>> newbie. Is the information easy to follow? On the flip side, if you
>>> wanted to help someone, ask yourself, if they follow the steps on  
>>> this
>>> page will that suffice? Are there places that need improvement? Can
>>> you think of anything else that can be added?
>>>
>>> NOTE: I'm still working on the TDM/OpenZAP section as well as the
>>> sections on scripting, event socket, elements of a jira ticket, etc.
>>> If you have suggestions for content on those pages please email me  
>>> off
>>> list or hop and and fill in some of the blanks.
>>>
>>> The core development team really appreciates all of your help. Now
>>> that FS is growing like mad we are at the point where it is  
>>> imperative
>>> that we have reliable documentation for new ones so that the
>>> developers and other experts can focus on advancing the project even
>>> further. Let's all lend a hand by improving the documentation.
>>>
>>> Thanks again!
>>> -MC (mercutioviz)
>>>
>>> _______________________________________________
>>> Freeswitch-dev mailing list
>>> Freeswitch-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> dev
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From seven at idapted.com  Wed Jan 21 20:30:23 2009
From: seven at idapted.com (seven du)
Date: Thu, 22 Jan 2009 12:30:23 +0800
Subject: [Freeswitch-dev] Recoding G729 raw payload in FreeSWITCH and G729
	decoding
Message-ID: <E03E6A42-445B-4EC8-BE7F-8B97885B0076@idapted.com>

Hi FreeSWITCHers,


1) FreeSWITCH support G729 codec in passthrough mode, it is normally  
engough if both call-legs suport G729. But there is no way to do  
recording if you can't decode it. I wrote a small module called  
mod_recpld. The idea is to record the raw payload in rtp packets to  
files, and decode them into 3rd party  converters.

See details at: http://code.google.com/p/mod-recpld/

2) EasyG729A implemented by http://imtelephone.com can be free of use  
on research purpose, I made a wrapper so it can be used in FreeSWITCH.

See deatila at: http://code.google.com/p/libg729/

The code is not good but just works.

BTW, I think it maybe easy to store codes on FreeSWITCH svn than on  
google code, how can I become a developer? Can I get a svn access on  
svn.freeswitch.org and own a branch?


From ludovic.fouquet at bewan.com  Thu Jan 22 02:43:42 2009
From: ludovic.fouquet at bewan.com (ludovic)
Date: Thu, 22 Jan 2009 11:43:42 +0100
Subject: [Freeswitch-dev] switch_frame structure question
In-Reply-To: <191c3a030901210954y41acc7c7r952d5021ddb5a5c8@mail.gmail.com>
References: <49775EE2.1060203@bewan.com>
	<191c3a030901210954y41acc7c7r952d5021ddb5a5c8@mail.gmail.com>
Message-ID: <49784DDE.5050509@bewan.com>

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From seven at idapted.com  Fri Jan 23 01:06:21 2009
From: seven at idapted.com (seven du)
Date: Fri, 23 Jan 2009 17:06:21 +0800
Subject: [Freeswitch-dev] mod_skypiax inching forward
Message-ID: <7CD31F58-816C-4CEA-BEA2-05AB5BE6770B@idapted.com>

hi Giovanni,

I just made some improve on skypiax, so you can call skypiax/ANY/ 
another_skypename, it will automatically chose an available channel.  
not good, but it works.

put the following code directly before the following line:


         for (i = 0;  i < SKYPIAX_MAX_INTERFACES; i++) {

And change the above line to

         for (i = 0; !found && i < SKYPIAX_MAX_INTERFACES; i++) {

It should be like:



		if (strncmp("ANY", interface_name,  strlen(interface_name)) == 0) {	
	      //find an available one, allowing call like originate skypiax/ 
ANY/another_skypename

		  DEBUGA_SKYPE("Finding one available skype interface\n",  
SKYPIAX_P_LOG);

	       for (i = 0; !found && i < SKYPIAX_MAX_INTERFACES; i++) {
       	   		if (strlen(globals.SKYPIAX_INTERFACES[i].name)) {
				  int skype_state = 0;

				  tech_pvt = &globals.SKYPIAX_INTERFACES[i];
				  skype_state = tech_pvt->interface_state;
				  DEBUGA_SKYPE("skype interface: %d, name: %s, state: %d\n",  
SKYPIAX_P_LOG,
				 	i, globals.SKYPIAX_INTERFACES[i].name, skype_state);
				  if (SKYPIAX_STATE_DOWN == skype_state || 0 == skype_state) {
				     found=1;
				     break;
		          }
   		      	}
		   	}
		}


         for (i = 0; !found && i < SKYPIAX_MAX_INTERFACES; i++) {

Regards,
Seven



From intralanman at freeswitch.org  Fri Jan 23 07:42:05 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Fri, 23 Jan 2009 09:42:05 -0600
Subject: [Freeswitch-dev] mod_skypiax inching forward
In-Reply-To: <7CD31F58-816C-4CEA-BEA2-05AB5BE6770B@idapted.com>
References: <7CD31F58-816C-4CEA-BEA2-05AB5BE6770B@idapted.com>
Message-ID: <4979E54D.5060903@freeswitch.org>

seven du wrote:
> hi Giovanni,
>
> I just made some improve on skypiax, so you can call skypiax/ANY/ 
> another_skypename, it will automatically chose an available channel.  
> not good, but it works.
>
> put the following code directly before the following line:
>   
the best way to submit improvements is to post a patch on jira. if you 
haven't yet, create an account there. svn diff > /tmp/my.patch should 
give you a nice patch to post on the tracker

-Ray


From msc at freeswitch.org  Fri Jan 23 08:00:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 23 Jan 2009 08:00:41 -0800
Subject: [Freeswitch-dev] mod_skypiax inching forward
In-Reply-To: <4979E54D.5060903@freeswitch.org>
References: <7CD31F58-816C-4CEA-BEA2-05AB5BE6770B@idapted.com>
	<4979E54D.5060903@freeswitch.org>
Message-ID: <87f2f3b90901230800q66111a62qfbd1ea683333f80e@mail.gmail.com>

Don't forget that all patches are text files and should have a .txt
ending and that diffs should always be done from the root of the
source tree. :)
-MC


On Fri, Jan 23, 2009 at 7:42 AM, Raymond Chandler
<intralanman at freeswitch.org> wrote:
> seven du wrote:
>> hi Giovanni,
>>
>> I just made some improve on skypiax, so you can call skypiax/ANY/
>> another_skypename, it will automatically chose an available channel.
>> not good, but it works.
>>
>> put the following code directly before the following line:
>>
> the best way to submit improvements is to post a patch on jira. if you
> haven't yet, create an account there. svn diff > /tmp/my.patch should
> give you a nice patch to post on the tracker
>
> -Ray
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From r.zagler at zakotel.com  Sat Jan 24 09:08:55 2009
From: r.zagler at zakotel.com (Roland Zagler)
Date: Sat, 24 Jan 2009 18:08:55 +0100
Subject: [Freeswitch-dev] Trouble registering SIP client
Message-ID: <850935DD0336FC4B98C006D9CC72B8154076CA@fog2k01.fog.rack>

Hi everybody,

I am writing a Java SIP ua (based on mjsip 1.6) that is supposed to
register to freeswitch.
Unfortunately I am not able to register and I cannot find the problem,
although a softphone
(X-Lite) CAN register by using the same creds. The credentials
are:

SIP username: roland
SIP password: roland
SIP realm: 10.0.4.60

There is no acl activated.

The freeswitch versions I have tested are 1.0.2 and the trunk of today
(11480).

I hope someone on the list could take a look on the trace I attached
below and give
me a hint what goes wrong.

Thank you very much in advance.

Roland




tport_wakeup_pri(0x81042c8): events IN
tport_recv_event(0x81042c8)
tport_recv_iovec(0x81042c8) msg 0xb3415ed8 from (udp/10.0.4.60:5060) has
373 bytes, veclen = 1
recv 373 bytes from udp/[10.0.2.51]:32816 at 17:01:30.461289:
 
------------------------------------------------------------------------
   REGISTER sip:10.0.4.60:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562
   Max-Forwards: 70
   To: "roland" <sip:roland at 10.0.4.60:5060>
   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
   Call-ID: 218108472980 at 10.0.2.51
   CSeq: 1 REGISTER
   Contact: <sip:roland at 10.0.2.51:32816>
   Expires: 3600
   User-Agent: ZaKoSIP v1.2
   Content-Length: 0
   
 
------------------------------------------------------------------------
tport_deliver(0x81042c8): msg 0xb3415ed8 (373 bytes) from
udp/10.0.2.51:5060/sip next=(nil)
nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 1)
nta: canonizing sip:10.0.4.60:5060 with contact
nta: REGISTER (1) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x80fe270, 0x80f9448, 0xb3402b30) called
soa_set_params(static::0xb341ba78, ...) called
nua: nua_application_event: entering
nua: nua_respond: entering
nua(0xb3402b30): sent signal r_respond
2009-01-24 18:01:30 [DEBUG] sofia_reg.c:883 sofia_reg_handle_register()
Requesting Registration from: [roland at 10.0.4.60]
nua: nua_handle_destroy: entering
nua(0xb3402b30): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
nua: nua_stack_set_params: entering
soa_set_params(static::0xb341ba78, ...) called
tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816
tport_resolve addrinfo = 10.0.2.51:32816
tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816
tport_vsend returned 620
send 620 bytes to udp/[10.0.2.51]:32816 at 17:01:30.586602:
 
------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562
   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
   To: "roland" <sip:roland at 10.0.4.60:5060>;tag=aZa7matyy5Uca
   Call-ID: 218108472980 at 10.0.2.51
   CSeq: 1 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="10.0.4.60",
nonce="a5608fcc-ea38-11dd-8437-6d678b851040", algorithm=MD5, qop="auth"
   Content-Length: 0
   
 
------------------------------------------------------------------------
nta: sent 401 Unauthorized for REGISTER (1)
nta: timer set to 32000 ms
nta_leg_destroy((nil))
soa_destroy(static::0xb341ba78) called
tport_wakeup_pri(0x81042c8): events IN
tport_recv_event(0x81042c8)
tport_recv_iovec(0x81042c8) msg 0xb3444050 from (udp/10.0.4.60:5060) has
575 bytes, veclen = 1
recv 575 bytes from udp/[10.0.2.51]:32816 at 17:01:30.614376:
 
------------------------------------------------------------------------
   REGISTER sip:10.0.4.60:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562
   Max-Forwards: 70
   To: "roland" <sip:roland at 10.0.4.60:5060>
   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
   Call-ID: 218108472980 at 10.0.2.51
   CSeq: 2 REGISTER
   Contact: <sip:roland at 10.0.2.51:32816>
   Expires: 3600
   User-Agent: ZaKoSIP v1.2
   Authorization: Digest username="roland", realm="10.0.4.60",
nonce="a5608fcc-ea38-11dd-8437-6d678b851040", uri="sip:10.0.4.60:5060",
algorithm=MD5, qop=auth, response="2f26d787c68ac2a342a4b1d7bb49a1a0"
   Content-Length: 0
   
 
------------------------------------------------------------------------
tport_deliver(0x81042c8): msg 0xb3444050 (575 bytes) from
udp/10.0.2.51:5060/sip next=(nil)
nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 2)
nta: canonizing sip:10.0.4.60:5060 with contact
nta: REGISTER (2) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x80fe270, 0x80f9448, 0xb3445690) called
soa_set_params(static::0xb341ba78, ...) called
nua: nua_application_event: entering
2009-01-24 18:01:30 [DEBUG] sofia_reg.c:869 sofia_reg_handle_register()
Send challenge for [roland at 10.0.4.60]
nua: nua_respond: entering
nua(0xb3445690): sent signal r_respond
nua: nua_handle_destroy: entering
nua(0xb3445690): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
nua: nua_stack_set_params: entering
soa_set_params(static::0xb341ba78, ...) called
tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816
tport_resolve addrinfo = 10.0.2.51:32816
tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816
tport_vsend returned 500
send 500 bytes to udp/[10.0.2.51]:32816 at 17:01:30.617125:
 
------------------------------------------------------------------------
   SIP/2.0 403 Forbidden
   Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562
   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
   To: "roland" <sip:roland at 10.0.4.60:5060>;tag=B83Zp5a2UejZN
   Call-ID: 218108472980 at 10.0.2.51
   CSeq: 2 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Content-Length: 0
   
 
------------------------------------------------------------------------
nta: sent 403 Forbidden for REGISTER (2)
nta_leg_destroy((nil))
soa_destroy(static::0xb341ba78) called



From mike at jerris.com  Sat Jan 24 11:19:20 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 24 Jan 2009 14:19:20 -0500
Subject: [Freeswitch-dev] Trouble registering SIP client
In-Reply-To: <850935DD0336FC4B98C006D9CC72B8154076CA@fog2k01.fog.rack>
References: <850935DD0336FC4B98C006D9CC72B8154076CA@fog2k01.fog.rack>
Message-ID: <3E6CCD32-B23A-4507-80A4-D3684CF170CF@jerris.com>

can you post the trace that works?  I have a feeling your setting up  
your auth hash that you are sending wrong, perhaps the code where you  
build that would help someone see the error.

Mike

On Jan 24, 2009, at 12:08 PM, Roland Zagler wrote:

> Hi everybody,
>
> I am writing a Java SIP ua (based on mjsip 1.6) that is supposed to
> register to freeswitch.
> Unfortunately I am not able to register and I cannot find the problem,
> although a softphone
> (X-Lite) CAN register by using the same creds. The credentials
> are:
>
> SIP username: roland
> SIP password: roland
> SIP realm: 10.0.4.60
>
> There is no acl activated.
>
> The freeswitch versions I have tested are 1.0.2 and the trunk of today
> (11480).
>
> I hope someone on the list could take a look on the trace I attached
> below and give
> me a hint what goes wrong.
>
> Thank you very much in advance.
>
> Roland
>
>
>
>
> tport_wakeup_pri(0x81042c8): events IN
> tport_recv_event(0x81042c8)
> tport_recv_iovec(0x81042c8) msg 0xb3415ed8 from (udp/10.0.4.60:5060)  
> has
> 373 bytes, veclen = 1
> recv 373 bytes from udp/[10.0.2.51]:32816 at 17:01:30.461289:
>
> ------------------------------------------------------------------------
>   REGISTER sip:10.0.4.60:5060 SIP/2.0
>   Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562
>   Max-Forwards: 70
>   To: "roland" <sip:roland at 10.0.4.60:5060>
>   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
>   Call-ID: 218108472980 at 10.0.2.51
>   CSeq: 1 REGISTER
>   Contact: <sip:roland at 10.0.2.51:32816>
>   Expires: 3600
>   User-Agent: ZaKoSIP v1.2
>   Content-Length: 0
>
>
> ------------------------------------------------------------------------
> tport_deliver(0x81042c8): msg 0xb3415ed8 (373 bytes) from
> udp/10.0.2.51:5060/sip next=(nil)
> nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 1)
> nta: canonizing sip:10.0.4.60:5060 with contact
> nta: REGISTER (1) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x80fe270, 0x80f9448, 0xb3402b30) called
> soa_set_params(static::0xb341ba78, ...) called
> nua: nua_application_event: entering
> nua: nua_respond: entering
> nua(0xb3402b30): sent signal r_respond
> 2009-01-24 18:01:30 [DEBUG] sofia_reg.c:883  
> sofia_reg_handle_register()
> Requesting Registration from: [roland at 10.0.4.60]
> nua: nua_handle_destroy: entering
> nua(0xb3402b30): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua: nua_stack_set_params: entering
> soa_set_params(static::0xb341ba78, ...) called
> tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816
> tport_resolve addrinfo = 10.0.2.51:32816
> tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816
> tport_vsend returned 620
> send 620 bytes to udp/[10.0.2.51]:32816 at 17:01:30.586602:
>
> ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562
>   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
>   To: "roland" <sip:roland at 10.0.4.60:5060>;tag=aZa7matyy5Uca
>   Call-ID: 218108472980 at 10.0.2.51
>   CSeq: 1 REGISTER
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   WWW-Authenticate: Digest realm="10.0.4.60",
> nonce="a5608fcc-ea38-11dd-8437-6d678b851040", algorithm=MD5,  
> qop="auth"
>   Content-Length: 0
>
>
> ------------------------------------------------------------------------
> nta: sent 401 Unauthorized for REGISTER (1)
> nta: timer set to 32000 ms
> nta_leg_destroy((nil))
> soa_destroy(static::0xb341ba78) called
> tport_wakeup_pri(0x81042c8): events IN
> tport_recv_event(0x81042c8)
> tport_recv_iovec(0x81042c8) msg 0xb3444050 from (udp/10.0.4.60:5060)  
> has
> 575 bytes, veclen = 1
> recv 575 bytes from udp/[10.0.2.51]:32816 at 17:01:30.614376:
>
> ------------------------------------------------------------------------
>   REGISTER sip:10.0.4.60:5060 SIP/2.0
>   Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562
>   Max-Forwards: 70
>   To: "roland" <sip:roland at 10.0.4.60:5060>
>   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
>   Call-ID: 218108472980 at 10.0.2.51
>   CSeq: 2 REGISTER
>   Contact: <sip:roland at 10.0.2.51:32816>
>   Expires: 3600
>   User-Agent: ZaKoSIP v1.2
>   Authorization: Digest username="roland", realm="10.0.4.60",
> nonce="a5608fcc-ea38-11dd-8437-6d678b851040", uri="sip: 
> 10.0.4.60:5060",
> algorithm=MD5, qop=auth, response="2f26d787c68ac2a342a4b1d7bb49a1a0"
>   Content-Length: 0
>
>
> ------------------------------------------------------------------------
> tport_deliver(0x81042c8): msg 0xb3444050 (575 bytes) from
> udp/10.0.2.51:5060/sip next=(nil)
> nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 2)
> nta: canonizing sip:10.0.4.60:5060 with contact
> nta: REGISTER (2) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x80fe270, 0x80f9448, 0xb3445690) called
> soa_set_params(static::0xb341ba78, ...) called
> nua: nua_application_event: entering
> 2009-01-24 18:01:30 [DEBUG] sofia_reg.c:869  
> sofia_reg_handle_register()
> Send challenge for [roland at 10.0.4.60]
> nua: nua_respond: entering
> nua(0xb3445690): sent signal r_respond
> nua: nua_handle_destroy: entering
> nua(0xb3445690): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua: nua_stack_set_params: entering
> soa_set_params(static::0xb341ba78, ...) called
> tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816
> tport_resolve addrinfo = 10.0.2.51:32816
> tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816
> tport_vsend returned 500
> send 500 bytes to udp/[10.0.2.51]:32816 at 17:01:30.617125:
>
> ------------------------------------------------------------------------
>   SIP/2.0 403 Forbidden
>   Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562
>   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
>   To: "roland" <sip:roland at 10.0.4.60:5060>;tag=B83Zp5a2UejZN
>   Call-ID: 218108472980 at 10.0.2.51
>   CSeq: 2 REGISTER
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   Content-Length: 0
>
>
> ------------------------------------------------------------------------
> nta: sent 403 Forbidden for REGISTER (2)
> nta_leg_destroy((nil))
> soa_destroy(static::0xb341ba78) called



From r.zagler at zakotel.com  Sat Jan 24 12:07:25 2009
From: r.zagler at zakotel.com (Roland Zagler)
Date: Sat, 24 Jan 2009 21:07:25 +0100
Subject: [Freeswitch-dev] Trouble registering SIP client
In-Reply-To: <3E6CCD32-B23A-4507-80A4-D3684CF170CF@jerris.com>
References: <850935DD0336FC4B98C006D9CC72B8154076CA@fog2k01.fog.rack>
	<3E6CCD32-B23A-4507-80A4-D3684CF170CF@jerris.com>
Message-ID: <850935DD0336FC4B98C006D9CC72B8154076CC@fog2k01.fog.rack>

Hi mike,

thanks for taking a look, here is the trace of a softphone called
firefly registering with the same creds,
the only difference is the ip the client runs on, I tested it already to
register from the
same ip, which didn't work. I also tried adding the "cnonce" and "nc"
parameters inside the authorization
header, also without success.

btw: I can register to an asterisk server using the same java client
code without probs.

thx again,
Roland



tport_wakeup_pri(0x80f3580): events IN
tport_recv_event(0x80f3580)
tport_recv_iovec(0x80f3580) msg 0xb34025c8 from (udp/10.0.4.60:5060) has
428 bytes, veclen = 1
recv 428 bytes from udp/[10.0.4.2]:5060 at 20:01:08.463700:
 
------------------------------------------------------------------------
   REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0
   To: "roland"<sip:roland at 10.0.4.60:5060;transport=udp>
   From: "roland"<sip:roland at 10.0.4.60>;tag=78399c75
   Via: SIP/2.0/UDP
10.0.4.2:5060;branch=z9hG4bK-d87543-0b5fac2e346e151d-1--d87543-;rport
   Call-ID: f85a4202fd310527 at YXBvbGxv
   CSeq: 1 REGISTER
   Contact: <sip:roland at 10.0.4.2:5060>;expires=7200
   Expires: 7200
   Max-Forwards: 70
   User-Agent: Firefly 2.0
   Content-Length: 0
   
 
------------------------------------------------------------------------
tport_deliver(0x80f3580): msg 0xb34025c8 (428 bytes) from
udp/10.0.4.2:5060/sip next=(nil)
nta: received REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0 (CSeq 1)
nta: canonizing sip:10.0.4.60:5060 with contact
nta: REGISTER (1) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x80f1740, 0x80eea10, 0x81258f0) called
soa_set_params(static::0x811b558, ...) called
nua: nua_application_event: entering
nua: nua_respond: entering
nua(0x81258f0): sent signal r_respond
2009-01-24 21:01:08 [DEBUG] sofia_reg.c:883 sofia_reg_handle_register()
Requesting Registration from: [roland at 10.0.4.60]
nua: nua_handle_destroy: entering
nua(0x81258f0): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
nua: nua_stack_set_params: entering
soa_set_params(static::0x811b558, ...) called
tport_tsend(0x80f3580) tpn = UDP/10.0.4.2:5060
tport_resolve addrinfo = 10.0.4.2:5060
tport_by_addrinfo(0x80f3580): not found by name UDP/10.0.4.2:5060
tport_vsend returned 651
send 651 bytes to udp/[10.0.4.2]:5060 at 20:01:08.550818:
 
------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP
10.0.4.2:5060;branch=z9hG4bK-d87543-0b5fac2e346e151d-1--d87543-;rport=50
60
   From: "roland"<sip:roland at 10.0.4.60>;tag=78399c75
   To: "roland"
<sip:roland at 10.0.4.60:5060;transport=udp>;tag=QmNHXB98gttgB
   Call-ID: f85a4202fd310527 at YXBvbGxv
   CSeq: 1 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="10.0.4.60",
nonce="bd912f5c-ea51-11dd-8d93-318210ee3033", algorithm=MD5, qop="auth"
   Content-Length: 0
   
 
------------------------------------------------------------------------
nta: sent 401 Unauthorized for REGISTER (1)
nta: timer set to 32000 ms
nta_leg_destroy((nil))
soa_destroy(static::0x811b558) called
tport_wakeup_pri(0x80f3580): events IN
tport_recv_event(0x80f3580)
tport_recv_iovec(0x80f3580) msg 0x812b358 from (udp/10.0.4.60:5060) has
668 bytes, veclen = 1
recv 668 bytes from udp/[10.0.4.2]:5060 at 20:01:08.553542:
 
------------------------------------------------------------------------
   REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0
   To: "roland"<sip:roland at 10.0.4.60:5060;transport=udp>
   From: "roland"<sip:roland at 10.0.4.60>;tag=78399c75
   Via: SIP/2.0/UDP
10.0.4.2:5060;branch=z9hG4bK-d87543-f16d4916bb41f15a-1--d87543-;rport
   Call-ID: f85a4202fd310527 at YXBvbGxv
   CSeq: 2 REGISTER
   Contact: <sip:roland at 10.0.4.2:5060>;expires=7200
   Expires: 7200
   Max-Forwards: 70
   User-Agent: Firefly 2.0
   Authorization: Digest
username="roland",realm="10.0.4.60",nonce="bd912f5c-ea51-11dd-8d93-31821
0ee3033",uri="sip:10.0.4.60:5060;transport=udp",response="f25e2b5a110923
4bae13b5976abe3129",cnonce="137481EB",nc=00000001,qop=auth,algorithm=MD5
   Content-Length: 0
   
 
------------------------------------------------------------------------
tport_deliver(0x80f3580): msg 0x812b358 (668 bytes) from
udp/10.0.4.2:5060/sip next=(nil)
nta: received REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0 (CSeq 2)
nta: canonizing sip:10.0.4.60:5060 with contact
nta: REGISTER (2) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x80f1740, 0x80eea10, 0xb342a660) called
soa_set_params(static::0xb342fc18, ...) called
nua: nua_application_event: entering
2009-01-24 21:01:08 [DEBUG] sofia_reg.c:971 sofia_reg_handle_register()
Register:
From:    [roland at 10.0.4.60]
Contact: ["roland" <sip:roland at 10.0.4.2:5060>]
Expires: [7200]
nua: nua_respond: entering
nua(0xb342a660): sent signal r_respond
nua: nua_handle_destroy: entering
nua(0xb342a660): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
nua: nua_stack_set_params: entering
soa_set_params(static::0xb342fc18, ...) called
tport_tsend(0x80f3580) tpn = UDP/10.0.4.2:5060
tport_resolve addrinfo = 10.0.4.2:5060
tport_by_addrinfo(0x80f3580): not found by name UDP/10.0.4.2:5060
tport_vsend returned 611
send 611 bytes to udp/[10.0.4.2]:5060 at 20:01:08.559893:
 
------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
10.0.4.2:5060;branch=z9hG4bK-d87543-f16d4916bb41f15a-1--d87543-;rport=50
60
   From: "roland"<sip:roland at 10.0.4.60>;tag=78399c75
   To: "roland"
<sip:roland at 10.0.4.60:5060;transport=udp>;tag=rXeaZ6Sce3g3p
   Call-ID: f85a4202fd310527 at YXBvbGxv
   CSeq: 2 REGISTER
   Contact: <sip:roland at 10.0.4.2:5060>;expires=7200
   Date: Sat, 24 Jan 2009 20:01:08 GMT
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Content-Length: 0
   
 
------------------------------------------------------------------------
nta: sent 200 OK for REGISTER (2)
nta_leg_destroy((nil))
soa_destroy(static::0xb342fc18) called

-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: Saturday, January 24, 2009 8:19 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] Trouble registering SIP client

can you post the trace that works?  I have a feeling your setting up  
your auth hash that you are sending wrong, perhaps the code where you  
build that would help someone see the error.

Mike

On Jan 24, 2009, at 12:08 PM, Roland Zagler wrote:

> Hi everybody,
>
> I am writing a Java SIP ua (based on mjsip 1.6) that is supposed to
> register to freeswitch.
> Unfortunately I am not able to register and I cannot find the problem,
> although a softphone
> (X-Lite) CAN register by using the same creds. The credentials
> are:
>
> SIP username: roland
> SIP password: roland
> SIP realm: 10.0.4.60
>
> There is no acl activated.
>
> The freeswitch versions I have tested are 1.0.2 and the trunk of today
> (11480).
>
> I hope someone on the list could take a look on the trace I attached
> below and give
> me a hint what goes wrong.
>
> Thank you very much in advance.
>
> Roland
>
>
>
>
> tport_wakeup_pri(0x81042c8): events IN
> tport_recv_event(0x81042c8)
> tport_recv_iovec(0x81042c8) msg 0xb3415ed8 from (udp/10.0.4.60:5060)  
> has
> 373 bytes, veclen = 1
> recv 373 bytes from udp/[10.0.2.51]:32816 at 17:01:30.461289:
>
>
------------------------------------------------------------------------
>   REGISTER sip:10.0.4.60:5060 SIP/2.0
>   Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562
>   Max-Forwards: 70
>   To: "roland" <sip:roland at 10.0.4.60:5060>
>   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
>   Call-ID: 218108472980 at 10.0.2.51
>   CSeq: 1 REGISTER
>   Contact: <sip:roland at 10.0.2.51:32816>
>   Expires: 3600
>   User-Agent: ZaKoSIP v1.2
>   Content-Length: 0
>
>
>
------------------------------------------------------------------------
> tport_deliver(0x81042c8): msg 0xb3415ed8 (373 bytes) from
> udp/10.0.2.51:5060/sip next=(nil)
> nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 1)
> nta: canonizing sip:10.0.4.60:5060 with contact
> nta: REGISTER (1) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x80fe270, 0x80f9448, 0xb3402b30) called
> soa_set_params(static::0xb341ba78, ...) called
> nua: nua_application_event: entering
> nua: nua_respond: entering
> nua(0xb3402b30): sent signal r_respond
> 2009-01-24 18:01:30 [DEBUG] sofia_reg.c:883  
> sofia_reg_handle_register()
> Requesting Registration from: [roland at 10.0.4.60]
> nua: nua_handle_destroy: entering
> nua(0xb3402b30): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua: nua_stack_set_params: entering
> soa_set_params(static::0xb341ba78, ...) called
> tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816
> tport_resolve addrinfo = 10.0.2.51:32816
> tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816
> tport_vsend returned 620
> send 620 bytes to udp/[10.0.2.51]:32816 at 17:01:30.586602:
>
>
------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562
>   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
>   To: "roland" <sip:roland at 10.0.4.60:5060>;tag=aZa7matyy5Uca
>   Call-ID: 218108472980 at 10.0.2.51
>   CSeq: 1 REGISTER
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   WWW-Authenticate: Digest realm="10.0.4.60",
> nonce="a5608fcc-ea38-11dd-8437-6d678b851040", algorithm=MD5,  
> qop="auth"
>   Content-Length: 0
>
>
>
------------------------------------------------------------------------
> nta: sent 401 Unauthorized for REGISTER (1)
> nta: timer set to 32000 ms
> nta_leg_destroy((nil))
> soa_destroy(static::0xb341ba78) called
> tport_wakeup_pri(0x81042c8): events IN
> tport_recv_event(0x81042c8)
> tport_recv_iovec(0x81042c8) msg 0xb3444050 from (udp/10.0.4.60:5060)  
> has
> 575 bytes, veclen = 1
> recv 575 bytes from udp/[10.0.2.51]:32816 at 17:01:30.614376:
>
>
------------------------------------------------------------------------
>   REGISTER sip:10.0.4.60:5060 SIP/2.0
>   Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562
>   Max-Forwards: 70
>   To: "roland" <sip:roland at 10.0.4.60:5060>
>   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
>   Call-ID: 218108472980 at 10.0.2.51
>   CSeq: 2 REGISTER
>   Contact: <sip:roland at 10.0.2.51:32816>
>   Expires: 3600
>   User-Agent: ZaKoSIP v1.2
>   Authorization: Digest username="roland", realm="10.0.4.60",
> nonce="a5608fcc-ea38-11dd-8437-6d678b851040", uri="sip: 
> 10.0.4.60:5060",
> algorithm=MD5, qop=auth, response="2f26d787c68ac2a342a4b1d7bb49a1a0"
>   Content-Length: 0
>
>
>
------------------------------------------------------------------------
> tport_deliver(0x81042c8): msg 0xb3444050 (575 bytes) from
> udp/10.0.2.51:5060/sip next=(nil)
> nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 2)
> nta: canonizing sip:10.0.4.60:5060 with contact
> nta: REGISTER (2) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x80fe270, 0x80f9448, 0xb3445690) called
> soa_set_params(static::0xb341ba78, ...) called
> nua: nua_application_event: entering
> 2009-01-24 18:01:30 [DEBUG] sofia_reg.c:869  
> sofia_reg_handle_register()
> Send challenge for [roland at 10.0.4.60]
> nua: nua_respond: entering
> nua(0xb3445690): sent signal r_respond
> nua: nua_handle_destroy: entering
> nua(0xb3445690): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua: nua_stack_set_params: entering
> soa_set_params(static::0xb341ba78, ...) called
> tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816
> tport_resolve addrinfo = 10.0.2.51:32816
> tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816
> tport_vsend returned 500
> send 500 bytes to udp/[10.0.2.51]:32816 at 17:01:30.617125:
>
>
------------------------------------------------------------------------
>   SIP/2.0 403 Forbidden
>   Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562
>   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
>   To: "roland" <sip:roland at 10.0.4.60:5060>;tag=B83Zp5a2UejZN
>   Call-ID: 218108472980 at 10.0.2.51
>   CSeq: 2 REGISTER
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   Content-Length: 0
>
>
>
------------------------------------------------------------------------
> nta: sent 403 Forbidden for REGISTER (2)
> nta_leg_destroy((nil))
> soa_destroy(static::0xb341ba78) called


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From anthony.minessale at gmail.com  Sat Jan 24 16:15:49 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 24 Jan 2009 18:15:49 -0600
Subject: [Freeswitch-dev] Trouble registering SIP client
In-Reply-To: <850935DD0336FC4B98C006D9CC72B8154076CC@fog2k01.fog.rack>
References: <850935DD0336FC4B98C006D9CC72B8154076CA@fog2k01.fog.rack>
	<3E6CCD32-B23A-4507-80A4-D3684CF170CF@jerris.com>
	<850935DD0336FC4B98C006D9CC72B8154076CC@fog2k01.fog.rack>
Message-ID: <191c3a030901241615t3ebce2b1r2bf32c7f19ad0a@mail.gmail.com>

consider this code:

This is what we do to auth: say your uri is sip:bar.com

first we compute uri digest which is the md5 hash of the string

REGISTER:sip:bar.com


Then depending on what you sent us we do another md5 hash with the following
info:




   if (nc && cnonce && qop) {
        input2 = switch_mprintf("%q:%q:%q:%q:%q:%q", a1_hash, nonce, nc,
cnonce, qop, uridigest);
    } else {
        input2 = switch_mprintf("%q:%q:%q", a1_hash, nonce, uridigest);
    }




On Sat, Jan 24, 2009 at 2:07 PM, Roland Zagler <r.zagler at zakotel.com> wrote:

> Hi mike,
>
> thanks for taking a look, here is the trace of a softphone called
> firefly registering with the same creds,
> the only difference is the ip the client runs on, I tested it already to
> register from the
> same ip, which didn't work. I also tried adding the "cnonce" and "nc"
> parameters inside the authorization
> header, also without success.
>
> btw: I can register to an asterisk server using the same java client
> code without probs.
>
> thx again,
> Roland
>
>
>
> tport_wakeup_pri(0x80f3580): events IN
> tport_recv_event(0x80f3580)
> tport_recv_iovec(0x80f3580) msg 0xb34025c8 from (udp/10.0.4.60:5060) has
> 428 bytes, veclen = 1
> recv 428 bytes from udp/[10.0.4.2]:5060 at 20:01:08.463700:
>
> ------------------------------------------------------------------------
>   REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0
>   To: "roland"<sip:roland at 10.0.4.60:5060;transport=udp>
>   From: "roland"<sip:roland at 10.0.4.60 <sip%3Aroland at 10.0.4.60>
> >;tag=78399c75
>    Via: SIP/2.0/UDP
> 10.0.4.2:5060;branch=z9hG4bK-d87543-0b5fac2e346e151d-1--d87543-;rport
>   Call-ID: f85a4202fd310527 at YXBvbGxv
>   CSeq: 1 REGISTER
>   Contact: <sip:roland at 10.0.4.2:5060>;expires=7200
>   Expires: 7200
>   Max-Forwards: 70
>   User-Agent: Firefly 2.0
>   Content-Length: 0
>
>
> ------------------------------------------------------------------------
> tport_deliver(0x80f3580): msg 0xb34025c8 (428 bytes) from
> udp/10.0.4.2:5060/sip next=(nil)
> nta: received REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0 (CSeq 1)
> nta: canonizing sip:10.0.4.60:5060 with contact
> nta: REGISTER (1) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x80f1740, 0x80eea10, 0x81258f0) called
> soa_set_params(static::0x811b558, ...) called
> nua: nua_application_event: entering
> nua: nua_respond: entering
> nua(0x81258f0): sent signal r_respond
> 2009-01-24 21:01:08 [DEBUG] sofia_reg.c:883 sofia_reg_handle_register()
> Requesting Registration from: [roland at 10.0.4.60]
> nua: nua_handle_destroy: entering
> nua(0x81258f0): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua: nua_stack_set_params: entering
> soa_set_params(static::0x811b558, ...) called
> tport_tsend(0x80f3580) tpn = UDP/10.0.4.2:5060
> tport_resolve addrinfo = 10.0.4.2:5060
> tport_by_addrinfo(0x80f3580): not found by name UDP/10.0.4.2:5060
> tport_vsend returned 651
> send 651 bytes to udp/[10.0.4.2]:5060 at 20:01:08.550818:
>
> ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   Via: SIP/2.0/UDP
> 10.0.4.2:5060;branch=z9hG4bK-d87543-0b5fac2e346e151d-1--d87543-;rport=50
> 60
>   From: "roland"<sip:roland at 10.0.4.60 <sip%3Aroland at 10.0.4.60>
> >;tag=78399c75
>   To: "roland"
> <sip:roland at 10.0.4.60:5060;transport=udp>;tag=QmNHXB98gttgB
>   Call-ID: f85a4202fd310527 at YXBvbGxv
>    CSeq: 1 REGISTER
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   WWW-Authenticate: Digest realm="10.0.4.60",
> nonce="bd912f5c-ea51-11dd-8d93-318210ee3033", algorithm=MD5, qop="auth"
>    Content-Length: 0
>
>
> ------------------------------------------------------------------------
> nta: sent 401 Unauthorized for REGISTER (1)
> nta: timer set to 32000 ms
> nta_leg_destroy((nil))
> soa_destroy(static::0x811b558) called
> tport_wakeup_pri(0x80f3580): events IN
> tport_recv_event(0x80f3580)
> tport_recv_iovec(0x80f3580) msg 0x812b358 from (udp/10.0.4.60:5060) has
> 668 bytes, veclen = 1
> recv 668 bytes from udp/[10.0.4.2]:5060 at 20:01:08.553542:
>
> ------------------------------------------------------------------------
>   REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0
>   To: "roland"<sip:roland at 10.0.4.60:5060;transport=udp>
>   From: "roland"<sip:roland at 10.0.4.60 <sip%3Aroland at 10.0.4.60>
> >;tag=78399c75
>    Via: SIP/2.0/UDP
> 10.0.4.2:5060;branch=z9hG4bK-d87543-f16d4916bb41f15a-1--d87543-;rport
>   Call-ID: f85a4202fd310527 at YXBvbGxv
>   CSeq: 2 REGISTER
>   Contact: <sip:roland at 10.0.4.2:5060>;expires=7200
>   Expires: 7200
>   Max-Forwards: 70
>   User-Agent: Firefly 2.0
>   Authorization: Digest
> username="roland",realm="10.0.4.60",nonce="bd912f5c-ea51-11dd-8d93-31821
> 0ee3033",uri="sip:10.0.4.60:5060;transport=udp",response="f25e2b5a110923
> 4bae13b5976abe3129",cnonce="137481EB",nc=00000001,qop=auth,algorithm=MD5
>   Content-Length: 0
>
>
> ------------------------------------------------------------------------
> tport_deliver(0x80f3580): msg 0x812b358 (668 bytes) from
> udp/10.0.4.2:5060/sip next=(nil)
> nta: received REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0 (CSeq 2)
> nta: canonizing sip:10.0.4.60:5060 with contact
> nta: REGISTER (2) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x80f1740, 0x80eea10, 0xb342a660) called
> soa_set_params(static::0xb342fc18, ...) called
> nua: nua_application_event: entering
> 2009-01-24 21:01:08 [DEBUG] sofia_reg.c:971 sofia_reg_handle_register()
> Register:
> From:    [roland at 10.0.4.60]
> Contact: ["roland" <sip:roland at 10.0.4.2:5060>]
> Expires: [7200]
> nua: nua_respond: entering
> nua(0xb342a660): sent signal r_respond
> nua: nua_handle_destroy: entering
> nua(0xb342a660): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua: nua_stack_set_params: entering
> soa_set_params(static::0xb342fc18, ...) called
> tport_tsend(0x80f3580) tpn = UDP/10.0.4.2:5060
> tport_resolve addrinfo = 10.0.4.2:5060
> tport_by_addrinfo(0x80f3580): not found by name UDP/10.0.4.2:5060
> tport_vsend returned 611
> send 611 bytes to udp/[10.0.4.2]:5060 at 20:01:08.559893:
>
> ------------------------------------------------------------------------
>   SIP/2.0 200 OK
>    Via: SIP/2.0/UDP
> 10.0.4.2:5060;branch=z9hG4bK-d87543-f16d4916bb41f15a-1--d87543-;rport=50
> 60
>   From: "roland"<sip:roland at 10.0.4.60 <sip%3Aroland at 10.0.4.60>
> >;tag=78399c75
>   To: "roland"
> <sip:roland at 10.0.4.60:5060;transport=udp>;tag=rXeaZ6Sce3g3p
>   Call-ID: f85a4202fd310527 at YXBvbGxv
>   CSeq: 2 REGISTER
>   Contact: <sip:roland at 10.0.4.2:5060>;expires=7200
>   Date: Sat, 24 Jan 2009 20:01:08 GMT
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>   Supported: timer, precondition, path, replaces
>   Content-Length: 0
>
>
> ------------------------------------------------------------------------
> nta: sent 200 OK for REGISTER (2)
> nta_leg_destroy((nil))
> soa_destroy(static::0xb342fc18) called
>
> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
> Michael Jerris
> Sent: Saturday, January 24, 2009 8:19 PM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] Trouble registering SIP client
>
> can you post the trace that works?  I have a feeling your setting up
> your auth hash that you are sending wrong, perhaps the code where you
> build that would help someone see the error.
>
> Mike
>
> On Jan 24, 2009, at 12:08 PM, Roland Zagler wrote:
>
> > Hi everybody,
> >
> > I am writing a Java SIP ua (based on mjsip 1.6) that is supposed to
> > register to freeswitch.
> > Unfortunately I am not able to register and I cannot find the problem,
> > although a softphone
> > (X-Lite) CAN register by using the same creds. The credentials
> > are:
> >
> > SIP username: roland
> > SIP password: roland
> > SIP realm: 10.0.4.60
> >
> > There is no acl activated.
> >
> > The freeswitch versions I have tested are 1.0.2 and the trunk of today
> > (11480).
> >
> > I hope someone on the list could take a look on the trace I attached
> > below and give
> > me a hint what goes wrong.
> >
> > Thank you very much in advance.
> >
> > Roland
> >
> >
> >
> >
> > tport_wakeup_pri(0x81042c8): events IN
> > tport_recv_event(0x81042c8)
> > tport_recv_iovec(0x81042c8) msg 0xb3415ed8 from (udp/10.0.4.60:5060)
> > has
> > 373 bytes, veclen = 1
> > recv 373 bytes from udp/[10.0.2.51]:32816 at 17:01:30.461289:
> >
> >
> ------------------------------------------------------------------------
> >   REGISTER sip:10.0.4.60:5060 SIP/2.0
> >   Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562
> >   Max-Forwards: 70
> >   To: "roland" <sip:roland at 10.0.4.60:5060>
> >   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
> >   Call-ID: 218108472980 at 10.0.2.51
> >   CSeq: 1 REGISTER
> >   Contact: <sip:roland at 10.0.2.51:32816>
> >   Expires: 3600
> >   User-Agent: ZaKoSIP v1.2
> >   Content-Length: 0
> >
> >
> >
> ------------------------------------------------------------------------
> > tport_deliver(0x81042c8): msg 0xb3415ed8 (373 bytes) from
> > udp/10.0.2.51:5060/sip next=(nil)
> > nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 1)
> > nta: canonizing sip:10.0.4.60:5060 with contact
> > nta: REGISTER (1) going to a default leg
> > nua: nua_stack_process_request: entering
> > nua: nh_create: entering
> > nua: nh_create_handle: entering
> > nua: nua_stack_set_params: entering
> > soa_clone(static::0x80fe270, 0x80f9448, 0xb3402b30) called
> > soa_set_params(static::0xb341ba78, ...) called
> > nua: nua_application_event: entering
> > nua: nua_respond: entering
> > nua(0xb3402b30): sent signal r_respond
> > 2009-01-24 18:01:30 [DEBUG] sofia_reg.c:883
> > sofia_reg_handle_register()
> > Requesting Registration from: [roland at 10.0.4.60]
> > nua: nua_handle_destroy: entering
> > nua(0xb3402b30): sent signal r_destroy
> > nua: nua_handle_magic: entering
> > nua: nua_handle_destroy: entering
> > nua: nua_stack_set_params: entering
> > soa_set_params(static::0xb341ba78, ...) called
> > tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816
> > tport_resolve addrinfo = 10.0.2.51:32816
> > tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816
> > tport_vsend returned 620
> > send 620 bytes to udp/[10.0.2.51]:32816 at 17:01:30.586602:
> >
> >
> ------------------------------------------------------------------------
> >   SIP/2.0 401 Unauthorized
> >   Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562
> >   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
> >   To: "roland" <sip:roland at 10.0.4.60:5060>;tag=aZa7matyy5Uca
> >   Call-ID: 218108472980 at 10.0.2.51
> >   CSeq: 1 REGISTER
> >   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
> >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
> >   Supported: timer, precondition, path, replaces
> >   WWW-Authenticate: Digest realm="10.0.4.60",
> > nonce="a5608fcc-ea38-11dd-8437-6d678b851040", algorithm=MD5,
> > qop="auth"
> >   Content-Length: 0
> >
> >
> >
> ------------------------------------------------------------------------
> > nta: sent 401 Unauthorized for REGISTER (1)
> > nta: timer set to 32000 ms
> > nta_leg_destroy((nil))
> > soa_destroy(static::0xb341ba78) called
> > tport_wakeup_pri(0x81042c8): events IN
> > tport_recv_event(0x81042c8)
> > tport_recv_iovec(0x81042c8) msg 0xb3444050 from (udp/10.0.4.60:5060)
> > has
> > 575 bytes, veclen = 1
> > recv 575 bytes from udp/[10.0.2.51]:32816 at 17:01:30.614376:
> >
> >
> ------------------------------------------------------------------------
> >   REGISTER sip:10.0.4.60:5060 SIP/2.0
> >   Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562
> >   Max-Forwards: 70
> >   To: "roland" <sip:roland at 10.0.4.60:5060>
> >   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
> >   Call-ID: 218108472980 at 10.0.2.51
> >   CSeq: 2 REGISTER
> >   Contact: <sip:roland at 10.0.2.51:32816>
> >   Expires: 3600
> >   User-Agent: ZaKoSIP v1.2
> >   Authorization: Digest username="roland", realm="10.0.4.60",
> > nonce="a5608fcc-ea38-11dd-8437-6d678b851040", uri="sip:
> > 10.0.4.60:5060",
> > algorithm=MD5, qop=auth, response="2f26d787c68ac2a342a4b1d7bb49a1a0"
> >   Content-Length: 0
> >
> >
> >
> ------------------------------------------------------------------------
> > tport_deliver(0x81042c8): msg 0xb3444050 (575 bytes) from
> > udp/10.0.2.51:5060/sip next=(nil)
> > nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 2)
> > nta: canonizing sip:10.0.4.60:5060 with contact
> > nta: REGISTER (2) going to a default leg
> > nua: nua_stack_process_request: entering
> > nua: nh_create: entering
> > nua: nh_create_handle: entering
> > nua: nua_stack_set_params: entering
> > soa_clone(static::0x80fe270, 0x80f9448, 0xb3445690) called
> > soa_set_params(static::0xb341ba78, ...) called
> > nua: nua_application_event: entering
> > 2009-01-24 18:01:30 [DEBUG] sofia_reg.c:869
> > sofia_reg_handle_register()
> > Send challenge for [roland at 10.0.4.60]
> > nua: nua_respond: entering
> > nua(0xb3445690): sent signal r_respond
> > nua: nua_handle_destroy: entering
> > nua(0xb3445690): sent signal r_destroy
> > nua: nua_handle_magic: entering
> > nua: nua_handle_destroy: entering
> > nua: nua_stack_set_params: entering
> > soa_set_params(static::0xb341ba78, ...) called
> > tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816
> > tport_resolve addrinfo = 10.0.2.51:32816
> > tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816
> > tport_vsend returned 500
> > send 500 bytes to udp/[10.0.2.51]:32816 at 17:01:30.617125:
> >
> >
> ------------------------------------------------------------------------
> >   SIP/2.0 403 Forbidden
> >   Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562
> >   From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
> >   To: "roland" <sip:roland at 10.0.4.60:5060>;tag=B83Zp5a2UejZN
> >   Call-ID: 218108472980 at 10.0.2.51
> >   CSeq: 2 REGISTER
> >   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
> >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
> >   Supported: timer, precondition, path, replaces
> >   Content-Length: 0
> >
> >
> >
> ------------------------------------------------------------------------
> > nta: sent 403 Forbidden for REGISTER (2)
> > nta_leg_destroy((nil))
> > soa_destroy(static::0xb341ba78) called
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From seven at idapted.com  Sun Jan 25 02:05:10 2009
From: seven at idapted.com (seven du)
Date: Sun, 25 Jan 2009 18:05:10 +0800
Subject: [Freeswitch-dev] mod_skypiax inching forward
In-Reply-To: <4979E54D.5060903@freeswitch.org>
References: <7CD31F58-816C-4CEA-BEA2-05AB5BE6770B@idapted.com>
	<4979E54D.5060903@freeswitch.org>
Message-ID: <5F53DE4B-AC72-491D-B1CC-3216BA2313FE@idapted.com>

Thanks.  I only had experience on post issues to jira. how to post a  
patch? also create an issure? And, the skypiax stuff is not in the  
trunk but a branch, is it also proper?

And I know the best way to make a patch is run svn diff in the root  
directory, but I only checked out the mod, but not the whole branch,  
so I think the best way to make a patch is run svn diff from the mod's  
root directory, is that right?


On Jan 23, 2009, at 11:42 PM, Raymond Chandler wrote:

> seven du wrote:
>> hi Giovanni,
>>
>> I just made some improve on skypiax, so you can call skypiax/ANY/
>> another_skypename, it will automatically chose an available channel.
>> not good, but it works.
>>
>> put the following code directly before the following line:
>>
> the best way to submit improvements is to post a patch on jira. if you
> haven't yet, create an account there. svn diff > /tmp/my.patch should
> give you a nice patch to post on the tracker
>
> -Ray
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org




From brian at freeswitch.org  Sun Jan 25 02:12:33 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 25 Jan 2009 04:12:33 -0600
Subject: [Freeswitch-dev] mod_skypiax inching forward
In-Reply-To: <5F53DE4B-AC72-491D-B1CC-3216BA2313FE@idapted.com>
References: <7CD31F58-816C-4CEA-BEA2-05AB5BE6770B@idapted.com>
	<4979E54D.5060903@freeswitch.org>
	<5F53DE4B-AC72-491D-B1CC-3216BA2313FE@idapted.com>
Message-ID: <F11CD300-33FF-4E24-A6C5-1A160A8466A2@freeswitch.org>

I have added mod_skypiax to Jira under endpoint modules and assigned  
the mod to Giovanni.  Please open a jira and attach any patches there.

Thanks,
/b

On Jan 25, 2009, at 4:05 AM, seven du wrote:

> Thanks.  I only had experience on post issues to jira. how to post a
> patch? also create an issure? And, the skypiax stuff is not in the
> trunk but a branch, is it also proper?
>
> And I know the best way to make a patch is run svn diff in the root
> directory, but I only checked out the mod, but not the whole branch,
> so I think the best way to make a patch is run svn diff from the mod's
> root directory, is that right?



From r.zagler at zakotel.com  Sun Jan 25 08:15:32 2009
From: r.zagler at zakotel.com (Roland Zagler)
Date: Sun, 25 Jan 2009 17:15:32 +0100
Subject: [Freeswitch-dev] Trouble registering SIP client [SOLVED]
In-Reply-To: <191c3a030901241615t3ebce2b1r2bf32c7f19ad0a@mail.gmail.com>
References: <850935DD0336FC4B98C006D9CC72B8154076CA@fog2k01.fog.rack><3E6CCD32-B23A-4507-80A4-D3684CF170CF@jerris.com><850935DD0336FC4B98C006D9CC72B8154076CC@fog2k01.fog.rack>
	<191c3a030901241615t3ebce2b1r2bf32c7f19ad0a@mail.gmail.com>
Message-ID: <850935DD0336FC4B98C006D9CC72B8154076CD@fog2k01.fog.rack>

Thank you Anthony,
this was the exact problem, the mjsip stack built a response field that contained the long version
(including "nc", "cnonce" and "qop") event if "nc" and "cnonce" were empty.

Thank you so much for your hint!

Roland




From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Sunday, January 25, 2009 1:16 AM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] Trouble registering SIP client

consider this code:

This is what we do to auth: say your uri is sip:bar.com

first we compute uri digest which is the md5 hash of the string 

REGISTER:sip:bar.com


Then depending on what you sent us we do another md5 hash with the following info:




?? if (nc && cnonce && qop) {
??????? input2 = switch_mprintf("%q:%q:%q:%q:%q:%q", a1_hash, nonce, nc, cnonce, qop, uridigest);
??? } else {
??????? input2 = switch_mprintf("%q:%q:%q", a1_hash, nonce, uridigest);
??? }



On Sat, Jan 24, 2009 at 2:07 PM, Roland Zagler <r.zagler at zakotel.com> wrote:
Hi mike,

thanks for taking a look, here is the trace of a softphone called
firefly registering with the same creds,
the only difference is the ip the client runs on, I tested it already to
register from the
same ip, which didn't work. I also tried adding the "cnonce" and "nc"
parameters inside the authorization
header, also without success.

btw: I can register to an asterisk server using the same java client
code without probs.

thx again,
Roland



tport_wakeup_pri(0x80f3580): events IN
tport_recv_event(0x80f3580)
tport_recv_iovec(0x80f3580) msg 0xb34025c8 from (udp/10.0.4.60:5060) has
428 bytes, veclen = 1
recv 428 bytes from udp/[10.0.4.2]:5060 at 20:01:08.463700:

------------------------------------------------------------------------
? REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0
? To: "roland"<sip:roland at 10.0.4.60:5060;transport=udp>
? From: "roland"<sip:roland at 10.0.4.60>;tag=78399c75
? Via: SIP/2.0/UDP
10.0.4.2:5060;branch=z9hG4bK-d87543-0b5fac2e346e151d-1--d87543-;rport
? Call-ID: f85a4202fd310527 at YXBvbGxv
? CSeq: 1 REGISTER
? Contact: <sip:roland at 10.0.4.2:5060>;expires=7200
? Expires: 7200
? Max-Forwards: 70
? User-Agent: Firefly 2.0
? Content-Length: 0


------------------------------------------------------------------------
tport_deliver(0x80f3580): msg 0xb34025c8 (428 bytes) from
udp/10.0.4.2:5060/sip next=(nil)
nta: received REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0 (CSeq 1)
nta: canonizing sip:10.0.4.60:5060 with contact
nta: REGISTER (1) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x80f1740, 0x80eea10, 0x81258f0) called
soa_set_params(static::0x811b558, ...) called
nua: nua_application_event: entering
nua: nua_respond: entering
nua(0x81258f0): sent signal r_respond
2009-01-24 21:01:08 [DEBUG] sofia_reg.c:883 sofia_reg_handle_register()
Requesting Registration from: [roland at 10.0.4.60]
nua: nua_handle_destroy: entering
nua(0x81258f0): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
nua: nua_stack_set_params: entering
soa_set_params(static::0x811b558, ...) called
tport_tsend(0x80f3580) tpn = UDP/10.0.4.2:5060
tport_resolve addrinfo = 10.0.4.2:5060
tport_by_addrinfo(0x80f3580): not found by name UDP/10.0.4.2:5060
tport_vsend returned 651
send 651 bytes to udp/[10.0.4.2]:5060 at 20:01:08.550818:

------------------------------------------------------------------------
? SIP/2.0 401 Unauthorized
? Via: SIP/2.0/UDP
10.0.4.2:5060;branch=z9hG4bK-d87543-0b5fac2e346e151d-1--d87543-;rport=50
60
? From: "roland"<sip:roland at 10.0.4.60>;tag=78399c75
? To: "roland"
<sip:roland at 10.0.4.60:5060;transport=udp>;tag=QmNHXB98gttgB
? Call-ID: f85a4202fd310527 at YXBvbGxv
? CSeq: 1 REGISTER
? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
? Supported: timer, precondition, path, replaces
? WWW-Authenticate: Digest realm="10.0.4.60",
nonce="bd912f5c-ea51-11dd-8d93-318210ee3033", algorithm=MD5, qop="auth"
? Content-Length: 0


------------------------------------------------------------------------
nta: sent 401 Unauthorized for REGISTER (1)
nta: timer set to 32000 ms
nta_leg_destroy((nil))
soa_destroy(static::0x811b558) called
tport_wakeup_pri(0x80f3580): events IN
tport_recv_event(0x80f3580)
tport_recv_iovec(0x80f3580) msg 0x812b358 from (udp/10.0.4.60:5060) has
668 bytes, veclen = 1
recv 668 bytes from udp/[10.0.4.2]:5060 at 20:01:08.553542:

------------------------------------------------------------------------
? REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0
? To: "roland"<sip:roland at 10.0.4.60:5060;transport=udp>
? From: "roland"<sip:roland at 10.0.4.60>;tag=78399c75
? Via: SIP/2.0/UDP
10.0.4.2:5060;branch=z9hG4bK-d87543-f16d4916bb41f15a-1--d87543-;rport
? Call-ID: f85a4202fd310527 at YXBvbGxv
? CSeq: 2 REGISTER
? Contact: <sip:roland at 10.0.4.2:5060>;expires=7200
? Expires: 7200
? Max-Forwards: 70
? User-Agent: Firefly 2.0
? Authorization: Digest
username="roland",realm="10.0.4.60",nonce="bd912f5c-ea51-11dd-8d93-31821
0ee3033",uri="sip:10.0.4.60:5060;transport=udp",response="f25e2b5a110923
4bae13b5976abe3129",cnonce="137481EB",nc=00000001,qop=auth,algorithm=MD5
? Content-Length: 0


------------------------------------------------------------------------
tport_deliver(0x80f3580): msg 0x812b358 (668 bytes) from
udp/10.0.4.2:5060/sip next=(nil)
nta: received REGISTER sip:10.0.4.60:5060;transport=udp SIP/2.0 (CSeq 2)
nta: canonizing sip:10.0.4.60:5060 with contact
nta: REGISTER (2) going to a default leg
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x80f1740, 0x80eea10, 0xb342a660) called
soa_set_params(static::0xb342fc18, ...) called
nua: nua_application_event: entering
2009-01-24 21:01:08 [DEBUG] sofia_reg.c:971 sofia_reg_handle_register()
Register:
From: ? ?[roland at 10.0.4.60]
Contact: ["roland" <sip:roland at 10.0.4.2:5060>]
Expires: [7200]
nua: nua_respond: entering
nua(0xb342a660): sent signal r_respond
nua: nua_handle_destroy: entering
nua(0xb342a660): sent signal r_destroy
nua: nua_handle_magic: entering
nua: nua_handle_destroy: entering
nua: nua_stack_set_params: entering
soa_set_params(static::0xb342fc18, ...) called
tport_tsend(0x80f3580) tpn = UDP/10.0.4.2:5060
tport_resolve addrinfo = 10.0.4.2:5060
tport_by_addrinfo(0x80f3580): not found by name UDP/10.0.4.2:5060
tport_vsend returned 611
send 611 bytes to udp/[10.0.4.2]:5060 at 20:01:08.559893:

------------------------------------------------------------------------
? SIP/2.0 200 OK
? Via: SIP/2.0/UDP
10.0.4.2:5060;branch=z9hG4bK-d87543-f16d4916bb41f15a-1--d87543-;rport=50
60
? From: "roland"<sip:roland at 10.0.4.60>;tag=78399c75
? To: "roland"
<sip:roland at 10.0.4.60:5060;transport=udp>;tag=rXeaZ6Sce3g3p
? Call-ID: f85a4202fd310527 at YXBvbGxv
? CSeq: 2 REGISTER
? Contact: <sip:roland at 10.0.4.2:5060>;expires=7200
? Date: Sat, 24 Jan 2009 20:01:08 GMT
? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
? Supported: timer, precondition, path, replaces
? Content-Length: 0


------------------------------------------------------------------------
nta: sent 200 OK for REGISTER (2)
nta_leg_destroy((nil))
soa_destroy(static::0xb342fc18) called

-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: Saturday, January 24, 2009 8:19 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] Trouble registering SIP client

can you post the trace that works? ?I have a feeling your setting up
your auth hash that you are sending wrong, perhaps the code where you
build that would help someone see the error.

Mike

On Jan 24, 2009, at 12:08 PM, Roland Zagler wrote:

> Hi everybody,
>
> I am writing a Java SIP ua (based on mjsip 1.6) that is supposed to
> register to freeswitch.
> Unfortunately I am not able to register and I cannot find the problem,
> although a softphone
> (X-Lite) CAN register by using the same creds. The credentials
> are:
>
> SIP username: roland
> SIP password: roland
> SIP realm: 10.0.4.60
>
> There is no acl activated.
>
> The freeswitch versions I have tested are 1.0.2 and the trunk of today
> (11480).
>
> I hope someone on the list could take a look on the trace I attached
> below and give
> me a hint what goes wrong.
>
> Thank you very much in advance.
>
> Roland
>
>
>
>
> tport_wakeup_pri(0x81042c8): events IN
> tport_recv_event(0x81042c8)
> tport_recv_iovec(0x81042c8) msg 0xb3415ed8 from (udp/10.0.4.60:5060)
> has
> 373 bytes, veclen = 1
> recv 373 bytes from udp/[10.0.2.51]:32816 at 17:01:30.461289:
>
>
------------------------------------------------------------------------
> ? REGISTER sip:10.0.4.60:5060 SIP/2.0
> ? Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562
> ? Max-Forwards: 70
> ? To: "roland" <sip:roland at 10.0.4.60:5060>
> ? From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
> ? Call-ID: 218108472980 at 10.0.2.51
> ? CSeq: 1 REGISTER
> ? Contact: <sip:roland at 10.0.2.51:32816>
> ? Expires: 3600
> ? User-Agent: ZaKoSIP v1.2
> ? Content-Length: 0
>
>
>
------------------------------------------------------------------------
> tport_deliver(0x81042c8): msg 0xb3415ed8 (373 bytes) from
> udp/10.0.2.51:5060/sip next=(nil)
> nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 1)
> nta: canonizing sip:10.0.4.60:5060 with contact
> nta: REGISTER (1) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x80fe270, 0x80f9448, 0xb3402b30) called
> soa_set_params(static::0xb341ba78, ...) called
> nua: nua_application_event: entering
> nua: nua_respond: entering
> nua(0xb3402b30): sent signal r_respond
> 2009-01-24 18:01:30 [DEBUG] sofia_reg.c:883
> sofia_reg_handle_register()
> Requesting Registration from: [roland at 10.0.4.60]
> nua: nua_handle_destroy: entering
> nua(0xb3402b30): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua: nua_stack_set_params: entering
> soa_set_params(static::0xb341ba78, ...) called
> tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816
> tport_resolve addrinfo = 10.0.2.51:32816
> tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816
> tport_vsend returned 620
> send 620 bytes to udp/[10.0.2.51]:32816 at 17:01:30.586602:
>
>
------------------------------------------------------------------------
> ? SIP/2.0 401 Unauthorized
> ? Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562
> ? From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
> ? To: "roland" <sip:roland at 10.0.4.60:5060>;tag=aZa7matyy5Uca
> ? Call-ID: 218108472980 at 10.0.2.51
> ? CSeq: 1 REGISTER
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
> ? Supported: timer, precondition, path, replaces
> ? WWW-Authenticate: Digest realm="10.0.4.60",
> nonce="a5608fcc-ea38-11dd-8437-6d678b851040", algorithm=MD5,
> qop="auth"
> ? Content-Length: 0
>
>
>
------------------------------------------------------------------------
> nta: sent 401 Unauthorized for REGISTER (1)
> nta: timer set to 32000 ms
> nta_leg_destroy((nil))
> soa_destroy(static::0xb341ba78) called
> tport_wakeup_pri(0x81042c8): events IN
> tport_recv_event(0x81042c8)
> tport_recv_iovec(0x81042c8) msg 0xb3444050 from (udp/10.0.4.60:5060)
> has
> 575 bytes, veclen = 1
> recv 575 bytes from udp/[10.0.2.51]:32816 at 17:01:30.614376:
>
>
------------------------------------------------------------------------
> ? REGISTER sip:10.0.4.60:5060 SIP/2.0
> ? Via: SIP/2.0/UDP 10.0.2.51:32816;rport;branch=z9hG4bK33562
> ? Max-Forwards: 70
> ? To: "roland" <sip:roland at 10.0.4.60:5060>
> ? From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
> ? Call-ID: 218108472980 at 10.0.2.51
> ? CSeq: 2 REGISTER
> ? Contact: <sip:roland at 10.0.2.51:32816>
> ? Expires: 3600
> ? User-Agent: ZaKoSIP v1.2
> ? Authorization: Digest username="roland", realm="10.0.4.60",
> nonce="a5608fcc-ea38-11dd-8437-6d678b851040", uri="sip:
> 10.0.4.60:5060",
> algorithm=MD5, qop=auth, response="2f26d787c68ac2a342a4b1d7bb49a1a0"
> ? Content-Length: 0
>
>
>
------------------------------------------------------------------------
> tport_deliver(0x81042c8): msg 0xb3444050 (575 bytes) from
> udp/10.0.2.51:5060/sip next=(nil)
> nta: received REGISTER sip:10.0.4.60:5060 SIP/2.0 (CSeq 2)
> nta: canonizing sip:10.0.4.60:5060 with contact
> nta: REGISTER (2) going to a default leg
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x80fe270, 0x80f9448, 0xb3445690) called
> soa_set_params(static::0xb341ba78, ...) called
> nua: nua_application_event: entering
> 2009-01-24 18:01:30 [DEBUG] sofia_reg.c:869
> sofia_reg_handle_register()
> Send challenge for [roland at 10.0.4.60]
> nua: nua_respond: entering
> nua(0xb3445690): sent signal r_respond
> nua: nua_handle_destroy: entering
> nua(0xb3445690): sent signal r_destroy
> nua: nua_handle_magic: entering
> nua: nua_handle_destroy: entering
> nua: nua_stack_set_params: entering
> soa_set_params(static::0xb341ba78, ...) called
> tport_tsend(0x81042c8) tpn = UDP/10.0.2.51:32816
> tport_resolve addrinfo = 10.0.2.51:32816
> tport_by_addrinfo(0x81042c8): not found by name UDP/10.0.2.51:32816
> tport_vsend returned 500
> send 500 bytes to udp/[10.0.2.51]:32816 at 17:01:30.617125:
>
>
------------------------------------------------------------------------
> ? SIP/2.0 403 Forbidden
> ? Via: SIP/2.0/UDP 10.0.2.51:32816;rport=32816;branch=z9hG4bK33562
> ? From: "roland" <sip:roland at 10.0.4.60:5060>;tag=z9hG4bK84430669
> ? To: "roland" <sip:roland at 10.0.4.60:5060>;tag=B83Zp5a2UejZN
> ? Call-ID: 218108472980 at 10.0.2.51
> ? CSeq: 2 REGISTER
> ? User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11480
> ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
> ? Supported: timer, precondition, path, replaces
> ? Content-Length: 0
>
>
>
------------------------------------------------------------------------
> nta: sent 403 Forbidden for REGISTER (2)
> nta_leg_destroy((nil))
> soa_destroy(static::0xb341ba78) called


_______________________________________________
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_______________________________________________
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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400


From gmaruzz at celliax.org  Mon Jan 26 01:21:10 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 26 Jan 2009 10:21:10 +0100
Subject: [Freeswitch-dev] mod_skypiax inching forward
In-Reply-To: <F11CD300-33FF-4E24-A6C5-1A160A8466A2@freeswitch.org>
References: <7CD31F58-816C-4CEA-BEA2-05AB5BE6770B@idapted.com>
	<4979E54D.5060903@freeswitch.org>
	<5F53DE4B-AC72-491D-B1CC-3216BA2313FE@idapted.com>
	<F11CD300-33FF-4E24-A6C5-1A160A8466A2@freeswitch.org>
Message-ID: <7b197bef0901260121n6fe2b5d9h5b72ab9681f252a8@mail.gmail.com>

Hi Brian and Seven,

nice to be in Jira, thanks Brian.

I'll be on call :-)!


Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Sun, Jan 25, 2009 at 11:12 AM, Brian West <brian at freeswitch.org> wrote:
> I have added mod_skypiax to Jira under endpoint modules and assigned
> the mod to Giovanni.  Please open a jira and attach any patches there.
>
> Thanks,
> /b
>
> On Jan 25, 2009, at 4:05 AM, seven du wrote:
>
>> Thanks.  I only had experience on post issues to jira. how to post a
>> patch? also create an issure? And, the skypiax stuff is not in the
>> trunk but a branch, is it also proper?
>>
>> And I know the best way to make a patch is run svn diff in the root
>> directory, but I only checked out the mod, but not the whole branch,
>> so I think the best way to make a patch is run svn diff from the mod's
>> root directory, is that right?
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From jgarland at jasongarland.com  Mon Jan 26 04:30:31 2009
From: jgarland at jasongarland.com (Jason Garland)
Date: Mon, 26 Jan 2009 07:30:31 -0500
Subject: [Freeswitch-dev] Recoding G729 raw payload in FreeSWITCH and
	G729 decoding
In-Reply-To: <E03E6A42-445B-4EC8-BE7F-8B97885B0076@idapted.com>
References: <E03E6A42-445B-4EC8-BE7F-8B97885B0076@idapted.com>
Message-ID: <5BB84C1D-9C33-44FD-A5E4-35C9D269B628@jasongarland.com>

I was just thinking about something like this the other day. My idea  
was to record calls as pcap files for debugging purposes. The ability  
to play back an RTP stream from within a pcap file could be very  
useful for reproducing customer issues. You could either record one  
RTP stream per pcap or stuff both streams into the pcap, and possibly  
even the sip signaling. I've used sipp to play back pcap files in the  
past to reproduce polycom bugs. You must give it a pcap file  
containing a single rtp stream.

Sent from my iPhone

On Jan 21, 2009, at 11:30 PM, seven du <seven at idapted.com> wrote:

> Hi FreeSWITCHers,
>
>
> 1) FreeSWITCH support G729 codec in passthrough mode, it is normally
> engough if both call-legs suport G729. But there is no way to do
> recording if you can't decode it. I wrote a small module called
> mod_recpld. The idea is to record the raw payload in rtp packets to
> files, and decode them into 3rd party  converters.
>
> See details at: http://code.google.com/p/mod-recpld/
>
> 2) EasyG729A implemented by http://imtelephone.com can be free of use
> on research purpose, I made a wrapper so it can be used in FreeSWITCH.
>
> See deatila at: http://code.google.com/p/libg729/
>
> The code is not good but just works.
>
> BTW, I think it maybe easy to store codes on FreeSWITCH svn than on
> google code, how can I become a developer? Can I get a svn access on
> svn.freeswitch.org and own a branch?
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org


From seven at idapted.com  Mon Jan 26 18:25:15 2009
From: seven at idapted.com (seven du)
Date: Tue, 27 Jan 2009 10:25:15 +0800
Subject: [Freeswitch-dev] Recoding G729 raw payload in FreeSWITCH and
	G729 decoding
In-Reply-To: <5BB84C1D-9C33-44FD-A5E4-35C9D269B628@jasongarland.com>
References: <E03E6A42-445B-4EC8-BE7F-8B97885B0076@idapted.com>
	<5BB84C1D-9C33-44FD-A5E4-35C9D269B628@jasongarland.com>
Message-ID: <43EB0D4F-26E5-489A-9A8C-43F98C965E0D@idapted.com>

Thank you Jason.

I think there is a tools called pcapsipdump can do the job you  
described.  Though
it is not in FreeSWITCH, I think it's enough for debugging purpose.  
The mod_recpld,
not a good name though, is written to tight with FS and give a  
standard command
line interface like "record".

BTW, I'm also thinking to record all RTP stream, for debugging and  
statistic purpose,
but haven't figured out how easy/hard to do that. The idea is just  
like that: http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-January/001789.html

Seven

On Jan 26, 2009, at 8:30 PM, Jason Garland wrote:

> I was just thinking about something like this the other day. My idea
> was to record calls as pcap files for debugging purposes. The ability
> to play back an RTP stream from within a pcap file could be very
> useful for reproducing customer issues. You could either record one
> RTP stream per pcap or stuff both streams into the pcap, and possibly
> even the sip signaling. I've used sipp to play back pcap files in the
> past to reproduce polycom bugs. You must give it a pcap file
> containing a single rtp stream.
>
> Sent from my iPhone
>
> On Jan 21, 2009, at 11:30 PM, seven du <seven at idapted.com> wrote:
>
>> Hi FreeSWITCHers,
>>
>>
>> 1) FreeSWITCH support G729 codec in passthrough mode, it is normally
>> engough if both call-legs suport G729. But there is no way to do
>> recording if you can't decode it. I wrote a small module called
>> mod_recpld. The idea is to record the raw payload in rtp packets to
>> files, and decode them into 3rd party  converters.
>>
>> See details at: http://code.google.com/p/mod-recpld/
>>
>> 2) EasyG729A implemented by http://imtelephone.com can be free of use
>> on research purpose, I made a wrapper so it can be used in  
>> FreeSWITCH.
>>
>> See deatila at: http://code.google.com/p/libg729/
>>
>> The code is not good but just works.
>>
>> BTW, I think it maybe easy to store codes on FreeSWITCH svn than on
>> google code, how can I become a developer? Can I get a svn access on
>> svn.freeswitch.org and own a branch?
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> dev
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From jgarland at jasongarland.com  Mon Jan 26 21:26:01 2009
From: jgarland at jasongarland.com (Jason Garland)
Date: Tue, 27 Jan 2009 00:26:01 -0500
Subject: [Freeswitch-dev] Recoding G729 raw payload in FreeSWITCH and
	G729 decoding
In-Reply-To: <43EB0D4F-26E5-489A-9A8C-43F98C965E0D@idapted.com>
References: <E03E6A42-445B-4EC8-BE7F-8B97885B0076@idapted.com>
	<5BB84C1D-9C33-44FD-A5E4-35C9D269B628@jasongarland.com>
	<43EB0D4F-26E5-489A-9A8C-43F98C965E0D@idapted.com>
Message-ID: <B99D4E3F-C2EB-4B62-991F-FA68B0F551DD@jasongarland.com>

I've looked at pcapsipdump and filed several bugs with that project on  
sourceforge. It only does UDP SIP, and only works on port 5060. It  
does not understand image or video codecs in the SDP.


Sent from my iPhone

On Jan 26, 2009, at 9:25 PM, seven du <seven at idapted.com> wrote:

> Thank you Jason.
>
> I think there is a tools called pcapsipdump can do the job you
> described.  Though
> it is not in FreeSWITCH, I think it's enough for debugging purpose.
> The mod_recpld,
> not a good name though, is written to tight with FS and give a
> standard command
> line interface like "record".
>
> BTW, I'm also thinking to record all RTP stream, for debugging and
> statistic purpose,
> but haven't figured out how easy/hard to do that. The idea is just
> like that: http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-January/001789.html
>
> Seven
>
> On Jan 26, 2009, at 8:30 PM, Jason Garland wrote:
>
>> I was just thinking about something like this the other day. My idea
>> was to record calls as pcap files for debugging purposes. The ability
>> to play back an RTP stream from within a pcap file could be very
>> useful for reproducing customer issues. You could either record one
>> RTP stream per pcap or stuff both streams into the pcap, and possibly
>> even the sip signaling. I've used sipp to play back pcap files in the
>> past to reproduce polycom bugs. You must give it a pcap file
>> containing a single rtp stream.
>>
>> Sent from my iPhone
>>
>> On Jan 21, 2009, at 11:30 PM, seven du <seven at idapted.com> wrote:
>>
>>> Hi FreeSWITCHers,
>>>
>>>
>>> 1) FreeSWITCH support G729 codec in passthrough mode, it is normally
>>> engough if both call-legs suport G729. But there is no way to do
>>> recording if you can't decode it. I wrote a small module called
>>> mod_recpld. The idea is to record the raw payload in rtp packets to
>>> files, and decode them into 3rd party  converters.
>>>
>>> See details at: http://code.google.com/p/mod-recpld/
>>>
>>> 2) EasyG729A implemented by http://imtelephone.com can be free of  
>>> use
>>> on research purpose, I made a wrapper so it can be used in
>>> FreeSWITCH.
>>>
>>> See deatila at: http://code.google.com/p/libg729/
>>>
>>> The code is not good but just works.
>>>
>>> BTW, I think it maybe easy to store codes on FreeSWITCH svn than on
>>> google code, how can I become a developer? Can I get a svn access on
>>> svn.freeswitch.org and own a branch?
>>>
>>> _______________________________________________
>>> Freeswitch-dev mailing list
>>> Freeswitch-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> dev
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> dev
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org


From matthew at matthew.at  Mon Jan 26 21:31:24 2009
From: matthew at matthew.at (Matthew Kaufman)
Date: Mon, 26 Jan 2009 21:31:24 -0800
Subject: [Freeswitch-dev] very early prototype of SLA/BLA
Message-ID: <497E9C2C.8040408@matthew.at>

See http://jira.freeswitch.org/browse/MODENDP-179 for how far I've 
gotten on doing SLA/BLA in the draft-anil-sipping-bla-02/03 style. If 
the patch is applied and the feature enabled and Polycom phones in use 
and the phones configured for type="shared" and thirdPartyName="<some 
extension>", the lights really work. And with a patch Anthony provided 
to me today to add support for INVITE w/REPLACES, you can even put a 
call on hold at one phone, go to another phone with a blinking light for 
that line, and pick it up. (But you can't pick up a second phone with 
the first still on the call and have them both join the call at the same 
time via an auto-created conference... yet)

Amazing how much work it takes to emulate an old key system.

Again, this is a total rough-draft hack of a prototype, just enough to 
get the lights working, and with at least one major memory leak and/or 
nua handle leak(s) that cause bad things to happen, so isn't for 
production at all yet. I will be continuing to work on it myself, but I 
know this is something that gets requests so I wanted to get the fact 
that I'm working on it out in the open ASAP.

Matthew Kaufman



From matthew at matthew.at  Mon Jan 26 21:57:38 2009
From: matthew at matthew.at (Matthew Kaufman)
Date: Mon, 26 Jan 2009 21:57:38 -0800
Subject: [Freeswitch-dev] very early prototype of SLA/BLA
In-Reply-To: <497E9C2C.8040408@matthew.at>
References: <497E9C2C.8040408@matthew.at>
Message-ID: <497EA252.8040305@matthew.at>

Matthew Kaufman wrote:
> See http://jira.freeswitch.org/browse/MODENDP-179 ...
Of course right after uploading that, I realized I hadn't copied over my 
fixes for the two hardcoded contact strings. There are two places that 
say SIPTAG_CONTACT_STR("<sla-agent at 198.202.199.1>") where it should 
actually be dynamically generated from the profile's identity rather 
than my hardcoded IP address.

However, jira.freeswitch.org isn't responding *and* I'm headed to bed 
now and going to my real job all day tomorrow, so if you try to play 
with this before I get an update in later this week, you'll run into 
this problem.

Matthew Kaufman


From jgarland at jasongarland.com  Tue Jan 27 06:51:23 2009
From: jgarland at jasongarland.com (Jason Garland)
Date: Tue, 27 Jan 2009 09:51:23 -0500
Subject: [Freeswitch-dev] very early prototype of SLA/BLA
In-Reply-To: <497EA252.8040305@matthew.at>
References: <497E9C2C.8040408@matthew.at> <497EA252.8040305@matthew.at>
Message-ID: <6E10C77C-69CC-407B-809F-F31D99C9A6CE@jasongarland.com>

This is great! You have no idea how long I've been waiting for  
something like this! Kudos! I can't wait to try it out. The conference  
feature will be icing on the cake!

- Jason

Sent from my iPhone

On Jan 27, 2009, at 12:57 AM, Matthew Kaufman <matthew at matthew.at>  
wrote:

> Matthew Kaufman wrote:
>> See http://jira.freeswitch.org/browse/MODENDP-179 ...
> Of course right after uploading that, I realized I hadn't copied  
> over my
> fixes for the two hardcoded contact strings. There are two places that
> say SIPTAG_CONTACT_STR("<sla-agent at 198.202.199.1>") where it should
> actually be dynamically generated from the profile's identity rather
> than my hardcoded IP address.
>
> However, jira.freeswitch.org isn't responding *and* I'm headed to bed
> now and going to my real job all day tomorrow, so if you try to play
> with this before I get an update in later this week, you'll run into
> this problem.
>
> Matthew Kaufman
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org


From janvb at live.com  Tue Jan 27 04:14:06 2009
From: janvb at live.com (Jan Berger)
Date: Tue, 27 Jan 2009 13:14:06 +0100
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <497EA252.8040305@matthew.at>
References: <497E9C2C.8040408@matthew.at> <497EA252.8040305@matthew.at>
Message-ID: <BAY117-W385E67CF1DE9E4B5A1D80CD2CB0@phx.gbl>


hi,
 
Do we have any plans for a SIGTRAN/SS7 stack on FreeSWITCH?
 
MTP/SCCP/ISUP/TCAP
 
Jan
 
_________________________________________________________________
News, entertainment and everything you care about at Live.com. Get it now!
http://www.live.com/getstarted.aspx
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From krice at freeswitch.org  Tue Jan 27 10:27:36 2009
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 27 Jan 2009 12:27:36 -0600
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <BAY117-W385E67CF1DE9E4B5A1D80CD2CB0@phx.gbl>
Message-ID: <C5A4AE38.4CAA6%krice@freeswitch.org>

There are things available for this in the commercial space.  One from
CometSig.com and one from sangoma.  Nothing at this time in the opensource
arena. (OpenSS7 will not be integrated with freeswitch due to licensing
issues)

Contact me off list if you are interested in either of the commercial
versions I can assist you with these things.

Ken



From: Jan Berger <janvb at live.com>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Tue, 27 Jan 2009 13:14:06 +0100
To: <freeswitch-dev at lists.freeswitch.org>
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack

hi,
 
Do we have any plans for a SIGTRAN/SS7 stack on FreeSWITCH?
 
MTP/SCCP/ISUP/TCAP
 
Jan
 


Get news, entertainment and everything you care about at Live.com. Check it
out! <http://www.live.com/getstarted.aspx >

_______________________________________________
Freeswitch-dev mailing list
Freeswitch-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
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From brian at freeswitch.org  Tue Jan 27 10:59:18 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Jan 2009 12:59:18 -0600
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <C5A4AE38.4CAA6%krice@freeswitch.org>
References: <C5A4AE38.4CAA6%krice@freeswitch.org>
Message-ID: <7A392E86-4649-46BA-80EF-EC36AFD48CC5@freeswitch.org>

We should have an open source one too thats MPL or BSD ... I don't  
agree with selling protocols like this commercially.  Guess thats just  
the Open Source in me wanting that stuff to be free.

/b

On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:

> There are things available for this in the commercial space.  One  
> from CometSig.com and one from sangoma.  Nothing at this time in the  
> opensource arena. (OpenSS7 will not be integrated with freeswitch  
> due to licensing issues)
>
> Contact me off list if you are interested in either of the  
> commercial versions I can assist you with these things.
>
> Ken

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From krice at freeswitch.org  Tue Jan 27 11:07:53 2009
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 27 Jan 2009 13:07:53 -0600
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <7A392E86-4649-46BA-80EF-EC36AFD48CC5@freeswitch.org>
Message-ID: <C5A4B7A9.4CABD%krice@freeswitch.org>

I don?t see a non Commercial one getting certification and then getting
widely used here in the states... That?s the problem you have to spend tons
of time to get them certified... Its not like SIP where they will pretty
much take anything and everything unfortunately

K



From: Brian West <brian at freeswitch.org>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Tue, 27 Jan 2009 12:59:18 -0600
To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

We should have an open source one too thats MPL or BSD ... I don't agree
with selling protocols like this commercially.  Guess thats just the Open
Source in me wanting that stuff to be free.

/b

On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:

> There are things available for this in the commercial space.  One from
> CometSig.com and one from sangoma.  Nothing at this time in the opensource
> arena. (OpenSS7 will not be integrated with freeswitch due to licensing
> issues)
> 
> Contact me off list if you are interested in either of the commercial versions
> I can assist you with these things.
> 
> Ken



_______________________________________________
Freeswitch-dev mailing list
Freeswitch-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
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From brian at freeswitch.org  Tue Jan 27 11:21:20 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 27 Jan 2009 13:21:20 -0600
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <C5A4B7A9.4CABD%krice@freeswitch.org>
References: <C5A4B7A9.4CABD%krice@freeswitch.org>
Message-ID: <8B709493-E01A-417D-BD75-0D5CADD92D8A@freeswitch.org>

Doesn't matter It should still be FREE!

/b

On Jan 27, 2009, at 1:07 PM, Ken Rice wrote:

> I don?t see a non Commercial one getting certification and then  
> getting widely used here in the states... That?s the problem you  
> have to spend tons of time to get them certified... Its not like SIP  
> where they will pretty much take anything and everything unfortunately
>
> K

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From mike at jerris.com  Tue Jan 27 12:06:11 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 27 Jan 2009 15:06:11 -0500
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <C5A4B7A9.4CABD%krice@freeswitch.org>
References: <C5A4B7A9.4CABD%krice@freeswitch.org>
Message-ID: <A2C1BF78-BE35-4594-BFC5-0A92D40BD1DE@jerris.com>

I still see a compelling business case for a foundation supported free  
open source set of libs for the base protocols, commercially supported  
by multiple contributors who build higher level applications using the  
base protocols.  In a situation like this, commercial sponsors can  
pool resources for certification.  Obviously anyone using a modified  
version would not be using certified code but the sponsors who pay for  
certification would still get multiple benefits.

Mike

On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:

> I don?t see a non Commercial one getting certification and then  
> getting widely used here in the states... That?s the problem you  
> have to spend tons of time to get them certified... Its not like SIP  
> where they will pretty much take anything and everything unfortunately
>
> K
>
> From: Brian West <brian at freeswitch.org>
> Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org 
> >
> Date: Tue, 27 Jan 2009 12:59:18 -0600
> To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org 
> >
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> We should have an open source one too thats MPL or BSD ... I don't  
> agree with selling protocols like this commercially.  Guess thats  
> just the Open Source in me wanting that stuff to be free.
>
> /b
>
> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
>
>> There are things available for this in the commercial space.  One  
>> from CometSig.com and one from sangoma.  Nothing at this time in  
>> the opensource arena. (OpenSS7 will not be integrated with  
>> freeswitch due to licensing issues)
>>
>> Contact me off list if you are interested in either of the  
>> commercial versions I can assist you with these things.
>>
>> Ken

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From janvb at live.com  Wed Jan 28 10:30:15 2009
From: janvb at live.com (Jan Berger)
Date: Wed, 28 Jan 2009 19:30:15 +0100
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <A2C1BF78-BE35-4594-BFC5-0A92D40BD1DE@jerris.com>
References: <C5A4B7A9.4CABD%krice@freeswitch.org>
	<A2C1BF78-BE35-4594-BFC5-0A92D40BD1DE@jerris.com>
Message-ID: <BAY123-W3004DF5437645FB7446091D2C80@phx.gbl>


This is a little like the hen and the egg...
 
No one will support you starting to write a SS7 ++ stack, but once you get going and the first stacks turns up, so will business.
 
Jan



From: mike at jerris.comTo: freeswitch-dev at lists.freeswitch.orgDate: Tue, 27 Jan 2009 15:06:11 -0500Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
I still see a compelling business case for a foundation supported free open source set of libs for the base protocols, commercially supported by multiple contributors who build higher level applications using the base protocols.  In a situation like this, commercial sponsors can pool resources for certification.  Obviously anyone using a modified version would not be using certified code but the sponsors who pay for certification would still get multiple benefits.

Mike

On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:

I don?t see a non Commercial one getting certification and then getting widely used here in the states... That?s the problem you have to spend tons of time to get them certified... Its not like SIP where they will pretty much take anything and everything unfortunatelyKFrom: Brian West <brian at freeswitch.org>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Tue, 27 Jan 2009 12:59:18 -0600To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackWe should have an open source one too thats MPL or BSD ... I don't agree with selling protocols like this commercially.  Guess thats just the Open Source in me wanting that stuff to be free./bOn Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
There are things available for this in the commercial space.  One from CometSig.com and one from sangoma.  Nothing at this time in the opensource arena. (OpenSS7 will not be integrated with freeswitch due to licensing issues)Contact me off list if you are interested in either of the commercial versions I can assist you with these things.Ken
_________________________________________________________________
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http://www.microsoft.com/windows/windowslive/
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From krice at freeswitch.org  Wed Jan 28 10:37:48 2009
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 28 Jan 2009 12:37:48 -0600
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <BAY123-W3004DF5437645FB7446091D2C80@phx.gbl>
Message-ID: <C5A6021C.4CD77%krice@freeswitch.org>

That?s the same thing we saw...  Everyone wants SS7 but no one wants to help
fund it... And to make matters worse just getting proper documentation for
ANSI SS7 is not exactly a cheap endeavor. Unfortunately telecordia and crew
still want to charge thousands of dollars for the docs when the ITU decided
ages ago to make those available for free... Even then its still a
substantial amount of effort to get something working (even if you have
active links you can use to help understand whats going on)

K



From: Jan Berger <janvb at live.com>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Wed, 28 Jan 2009 19:30:15 +0100
To: <freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

This is a little like the hen and the egg...
 
No one will support you starting to write a SS7 ++ stack, but once you get
going and the first stacks turns up, so will business.
 
Jan



From: mike at jerris.com
To: freeswitch-dev at lists.freeswitch.org
Date: Tue, 27 Jan 2009 15:06:11 -0500
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack


I still see a compelling business case for a foundation supported free open
source set of libs for the base protocols, commercially supported by
multiple contributors who build higher level applications using the base
protocols.  In a situation like this, commercial sponsors can pool resources
for certification.  Obviously anyone using a modified version would not be
using certified code but the sponsors who pay for certification would still
get multiple benefits.

Mike

On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:

> I don?t see a non Commercial one getting certification and then getting widely
> used here in the states... That?s the problem you have to spend tons of time
> to get them certified... Its not like SIP where they will pretty much take
> anything and everything unfortunately
> 
> K
> 
> From: Brian West <brian at freeswitch.org>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Tue, 27 Jan 2009 12:59:18 -0600
> To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> 
> We should have an open source one too thats MPL or BSD ... I don't agree with
> selling protocols like this commercially.  Guess thats just the Open Source in
> me wanting that stuff to be free.
> 
> /b
> 
> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
> 
>> There are things available for this in the commercial space.  One from
>> CometSig.com and one from sangoma.  Nothing at this time in the opensource
>> arena. (OpenSS7 will not be integrated with freeswitch due to licensing
>> issues)
>> 
>> Contact me off list if you are interested in either of the commercial
>> versions I can assist you with these things.
>> 
>> Ken
> 
> 
> 
> check out the rest of the Windows Live?. More than mail?Windows Live? goes way
> beyond your inbox. More than messages
> <http://www.microsoft.com/windows/windowslive/>
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From janvb at live.com  Wed Jan 28 11:22:38 2009
From: janvb at live.com (Jan Berger)
Date: Wed, 28 Jan 2009 20:22:38 +0100
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <C5A6021C.4CD77%krice@freeswitch.org>
References: <BAY123-W3004DF5437645FB7446091D2C80@phx.gbl>
	<C5A6021C.4CD77%krice@freeswitch.org>
Message-ID: <BAY125-W27F7B5DA0459D16F097397D2C80@phx.gbl>


I live in Europe - and ITU/ETSI are free :)
 



Date: Wed, 28 Jan 2009 12:37:48 -0600From: krice at freeswitch.orgTo: freeswitch-dev at lists.freeswitch.orgSubject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackThat?s the same thing we saw...  Everyone wants SS7 but no one wants to help fund it... And to make matters worse just getting proper documentation for ANSI SS7 is not exactly a cheap endeavor. Unfortunately telecordia and crew still want to charge thousands of dollars for the docs when the ITU decided ages ago to make those available for free... Even then its still a substantial amount of effort to get something working (even if you have active links you can use to help understand whats going on)K



From: Jan Berger <janvb at live.com>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Wed, 28 Jan 2009 19:30:15 +0100To: <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackThis is a little like the hen and the egg... No one will support you starting to write a SS7 ++ stack, but once you get going and the first stacks turns up, so will business. Jan



From: mike at jerris.comTo: freeswitch-dev at lists.freeswitch.orgDate: Tue, 27 Jan 2009 15:06:11 -0500Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackI still see a compelling business case for a foundation supported free open source set of libs for the base protocols, commercially supported by multiple contributors who build higher level applications using the base protocols.  In a situation like this, commercial sponsors can pool resources for certification.  Obviously anyone using a modified version would not be using certified code but the sponsors who pay for certification would still get multiple benefits.MikeOn Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
I don?t see a non Commercial one getting certification and then getting widely used here in the states... That?s the problem you have to spend tons of time to get them certified... Its not like SIP where they will pretty much take anything and everything unfortunatelyKFrom: Brian West <brian at freeswitch.org>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Tue, 27 Jan 2009 12:59:18 -0600To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackWe should have an open source one too thats MPL or BSD ... I don't agree with selling protocols like this commercially.  Guess thats just the Open Source in me wanting that stuff to be free./bOn Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
There are things available for this in the commercial space.  One from CometSig.com and one from sangoma.  Nothing at this time in the opensource arena. (OpenSS7 will not be integrated with freeswitch due to licensing issues)Contact me off list if you are interested in either of the commercial versions I can assist you with these things.Ken

check out the rest of the Windows Live?. More than mail?Windows Live? goes way beyond your inbox. More than messages <http://www.microsoft.com/windows/windowslive/> 

_______________________________________________Freeswitch-dev mailing listFreeswitch-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-devUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
_________________________________________________________________
More than messages?check out the rest of the Windows Live?.
http://www.microsoft.com/windows/windowslive/
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From krice at freeswitch.org  Wed Jan 28 11:36:02 2009
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 28 Jan 2009 13:36:02 -0600
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <BAY125-W27F7B5DA0459D16F097397D2C80@phx.gbl>
Message-ID: <C5A60FC2.4CD8A%krice@freeswitch.org>

Must be nice ;) 

Here in the states they want you to pay for documentation on cable pinouts
and don?t even think about getting documentation for something like TCAP
LIDB without giving up your first born and an arm.



From: Jan Berger <janvb at live.com>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Wed, 28 Jan 2009 20:22:38 +0100
To: <freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

I live in Europe - and ITU/ETSI are free :)
 


Date: Wed, 28 Jan 2009 12:37:48 -0600
From: krice at freeswitch.org
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

That?s the same thing we saw...  Everyone wants SS7 but no one wants to help
fund it... And to make matters worse just getting proper documentation for
ANSI SS7 is not exactly a cheap endeavor. Unfortunately telecordia and crew
still want to charge thousands of dollars for the docs when the ITU decided
ages ago to make those available for free... Even then its still a
substantial amount of effort to get something working (even if you have
active links you can use to help understand whats going on)

K





From: Jan Berger <janvb at live.com>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Wed, 28 Jan 2009 19:30:15 +0100
To: <freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

This is a little like the hen and the egg...
 
No one will support you starting to write a SS7 ++ stack, but once you get
going and the first stacks turns up, so will business.
 
Jan





From: mike at jerris.com
To: freeswitch-dev at lists.freeswitch.org
Date: Tue, 27 Jan 2009 15:06:11 -0500
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack


I still see a compelling business case for a foundation supported free open
source set of libs for the base protocols, commercially supported by
multiple contributors who build higher level applications using the base
protocols.  In a situation like this, commercial sponsors can pool resources
for certification.  Obviously anyone using a modified version would not be
using certified code but the sponsors who pay for certification would still
get multiple benefits.

Mike

On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:


> I don?t see a non Commercial one getting certification and then getting widely
> used here in the states... That?s the problem you have to spend tons of time
> to get them certified... Its not like SIP where they will pretty much take
> anything and everything unfortunately
> 
> K
> 
> From: Brian West <brian at freeswitch.org>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Tue, 27 Jan 2009 12:59:18 -0600
> To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> 
> We should have an open source one too thats MPL or BSD ... I don't agree with
> selling protocols like this commercially.  Guess thats just the Open Source in
> me wanting that stuff to be free.
> 
> /b
> 
> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
> 
>> There are things available for this in the commercial space.  One from
>> CometSig.com and one from sangoma.  Nothing at this time in the opensource
>> arena. (OpenSS7 will not be integrated with freeswitch due to licensing
>> issues)
>> 
>> Contact me off list if you are interested in either of the commercial
>> versions I can assist you with these things.
>> 
>> Ken
> 
> 
> 
> check out the rest of the Windows Live?. More than mail?Windows Live? goes way
> beyond your inbox. More than messages
> <http://www.microsoft.com/windows/windowslive/>
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> check out the rest of the Windows Live?. More than mail?Windows Live? goes way
> beyond your inbox. More than messages
> <http://www.microsoft.com/windows/windowslive/>
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From janvb at live.com  Wed Jan 28 11:42:07 2009
From: janvb at live.com (Jan Berger)
Date: Wed, 28 Jan 2009 20:42:07 +0100
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <C5A60FC2.4CD8A%krice@freeswitch.org>
References: <BAY125-W27F7B5DA0459D16F097397D2C80@phx.gbl>
	<C5A60FC2.4CD8A%krice@freeswitch.org>
Message-ID: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>


Next time try to search if AT&T have a free version of the doc first - you might get a suprice.



Date: Wed, 28 Jan 2009 13:36:02 -0600From: krice at freeswitch.orgTo: freeswitch-dev at lists.freeswitch.orgSubject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackMust be nice ;) Here in the states they want you to pay for documentation on cable pinouts and don?t even think about getting documentation for something like TCAP LIDB without giving up your first born and an arm.



From: Jan Berger <janvb at live.com>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Wed, 28 Jan 2009 20:22:38 +0100To: <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackI live in Europe - and ITU/ETSI are free :) 



Date: Wed, 28 Jan 2009 12:37:48 -0600From: krice at freeswitch.orgTo: freeswitch-dev at lists.freeswitch.orgSubject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackThat?s the same thing we saw...  Everyone wants SS7 but no one wants to help fund it... And to make matters worse just getting proper documentation for ANSI SS7 is not exactly a cheap endeavor. Unfortunately telecordia and crew still want to charge thousands of dollars for the docs when the ITU decided ages ago to make those available for free... Even then its still a substantial amount of effort to get something working (even if you have active links you can use to help understand whats going on)K



From: Jan Berger <janvb at live.com>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Wed, 28 Jan 2009 19:30:15 +0100To: <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackThis is a little like the hen and the egg... No one will support you starting to write a SS7 ++ stack, but once you get going and the first stacks turns up, so will business. Jan



From: mike at jerris.comTo: freeswitch-dev at lists.freeswitch.orgDate: Tue, 27 Jan 2009 15:06:11 -0500Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackI still see a compelling business case for a foundation supported free open source set of libs for the base protocols, commercially supported by multiple contributors who build higher level applications using the base protocols.  In a situation like this, commercial sponsors can pool resources for certification.  Obviously anyone using a modified version would not be using certified code but the sponsors who pay for certification would still get multiple benefits.MikeOn Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
I don?t see a non Commercial one getting certification and then getting widely used here in the states... That?s the problem you have to spend tons of time to get them certified... Its not like SIP where they will pretty much take anything and everything unfortunatelyKFrom: Brian West <brian at freeswitch.org>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Tue, 27 Jan 2009 12:59:18 -0600To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackWe should have an open source one too thats MPL or BSD ... I don't agree with selling protocols like this commercially.  Guess thats just the Open Source in me wanting that stuff to be free./bOn Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
There are things available for this in the commercial space.  One from CometSig.com and one from sangoma.  Nothing at this time in the opensource arena. (OpenSS7 will not be integrated with freeswitch due to licensing issues)Contact me off list if you are interested in either of the commercial versions I can assist you with these things.Ken

check out the rest of the Windows Live?. More than mail?Windows Live? goes way beyond your inbox. More than messages <http://www.microsoft.com/windows/windowslive/> 

_______________________________________________Freeswitch-dev mailing listFreeswitch-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-devUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org <http://www.freeswitch.org/> 

check out the rest of the Windows Live?. More than mail?Windows Live? goes way beyond your inbox. More than messages <http://www.microsoft.com/windows/windowslive/> 

_______________________________________________Freeswitch-dev mailing listFreeswitch-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-devUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
_________________________________________________________________
News, entertainment and everything you care about at Live.com. Get it now!
http://www.live.com/getstarted.aspx
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From krice at freeswitch.org  Wed Jan 28 11:45:07 2009
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 28 Jan 2009 13:45:07 -0600
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
Message-ID: <C5A611E3.4CD8D%krice@freeswitch.org>

We did haha... Searched high and low... Ended up reverse engineering part of
the spec while we tried to come up with documentation...



From: Jan Berger <janvb at live.com>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Wed, 28 Jan 2009 20:42:07 +0100
To: <freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

Next time try to search if AT&T have a free version of the doc first - you
might get a suprice.



Date: Wed, 28 Jan 2009 13:36:02 -0600
From: krice at freeswitch.org
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

Must be nice ;) 

Here in the states they want you to pay for documentation on cable pinouts
and don?t even think about getting documentation for something like TCAP
LIDB without giving up your first born and an arm.





From: Jan Berger <janvb at live.com>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Wed, 28 Jan 2009 20:22:38 +0100
To: <freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

I live in Europe - and ITU/ETSI are free :)
 




Date: Wed, 28 Jan 2009 12:37:48 -0600
From: krice at freeswitch.org
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

That?s the same thing we saw...  Everyone wants SS7 but no one wants to help
fund it... And to make matters worse just getting proper documentation for
ANSI SS7 is not exactly a cheap endeavor. Unfortunately telecordia and crew
still want to charge thousands of dollars for the docs when the ITU decided
ages ago to make those available for free... Even then its still a
substantial amount of effort to get something working (even if you have
active links you can use to help understand whats going on)

K







From: Jan Berger <janvb at live.com>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Wed, 28 Jan 2009 19:30:15 +0100
To: <freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

This is a little like the hen and the egg...
 
No one will support you starting to write a SS7 ++ stack, but once you get
going and the first stacks turns up, so will business.
 
Jan







From: mike at jerris.com
To: freeswitch-dev at lists.freeswitch.org
Date: Tue, 27 Jan 2009 15:06:11 -0500
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack


I still see a compelling business case for a foundation supported free open
source set of libs for the base protocols, commercially supported by
multiple contributors who build higher level applications using the base
protocols.  In a situation like this, commercial sponsors can pool resources
for certification.  Obviously anyone using a modified version would not be
using certified code but the sponsors who pay for certification would still
get multiple benefits.

Mike

On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:



> I don?t see a non Commercial one getting certification and then getting widely
> used here in the states... That?s the problem you have to spend tons of time
> to get them certified... Its not like SIP where they will pretty much take
> anything and everything unfortunately
> 
> K
> 
> From: Brian West <brian at freeswitch.org>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Tue, 27 Jan 2009 12:59:18 -0600
> To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> 
> We should have an open source one too thats MPL or BSD ... I don't agree with
> selling protocols like this commercially.  Guess thats just the Open Source in
> me wanting that stuff to be free.
> 
> /b
> 
> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
> 
>> There are things available for this in the commercial space.  One from
>> CometSig.com and one from sangoma.  Nothing at this time in the opensource
>> arena. (OpenSS7 will not be integrated with freeswitch due to licensing
>> issues)
>> 
>> Contact me off list if you are interested in either of the commercial
>> versions I can assist you with these things.
>> 
>> Ken
> 
> 
> 
> check out the rest of the Windows Live?. More than mail?Windows Live? goes way
> beyond your inbox. More than messages
> <http://www.microsoft.com/windows/windowslive/>
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/>
> <http://www.freeswitch.org/>
> 
> 
> check out the rest of the Windows Live?. More than mail?Windows Live? goes way
> beyond your inbox. More than messages
> <http://www.microsoft.com/windows/windowslive/>
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> Get news, entertainment and everything you care about at Live.com. Check it
> out! <http://www.live.com/getstarted.aspx >
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From janvb at live.com  Wed Jan 28 11:53:52 2009
From: janvb at live.com (Jan Berger)
Date: Wed, 28 Jan 2009 20:53:52 +0100
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <C5A611E3.4CD8D%krice@freeswitch.org>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<C5A611E3.4CD8D%krice@freeswitch.org>
Message-ID: <BAY119-W3ABD90DFEA06701EE0292D2C80@phx.gbl>


"We" is that SS7box ?

Date: Wed, 28 Jan 2009 13:45:07 -0600From: krice at freeswitch.orgTo: freeswitch-dev at lists.freeswitch.orgSubject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackWe did haha... Searched high and low... Ended up reverse engineering part of the spec while we tried to come up with documentation... 

From: Jan Berger <janvb at live.com>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Wed, 28 Jan 2009 20:42:07 +0100To: <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackNext time try to search if AT&T have a free version of the doc first - you might get a suprice.

Date: Wed, 28 Jan 2009 13:36:02 -0600From: krice at freeswitch.orgTo: freeswitch-dev at lists.freeswitch.orgSubject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackMust be nice ;) Here in the states they want you to pay for documentation on cable pinouts and don?t even think about getting documentation for something like TCAP LIDB without giving up your first born and an arm.

From: Jan Berger <janvb at live.com>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Wed, 28 Jan 2009 20:22:38 +0100To: <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackI live in Europe - and ITU/ETSI are free :) 

Date: Wed, 28 Jan 2009 12:37:48 -0600From: krice at freeswitch.orgTo: freeswitch-dev at lists.freeswitch.orgSubject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackThat?s the same thing we saw...  Everyone wants SS7 but no one wants to help fund it... And to make matters worse just getting proper documentation for ANSI SS7 is not exactly a cheap endeavor. Unfortunately telecordia and crew still want to charge thousands of dollars for the docs when the ITU decided ages ago to make those available for free... Even then its still a substantial amount of effort to get something working (even if you have active links you can use to help understand whats going on)K

From: Jan Berger <janvb at live.com>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Wed, 28 Jan 2009 19:30:15 +0100To: <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackThis is a little like the hen and the egg... No one will support you starting to write a SS7 ++ stack, but once you get going and the first stacks turns up, so will business. Jan

From: mike at jerris.comTo: freeswitch-dev at lists.freeswitch.orgDate: Tue, 27 Jan 2009 15:06:11 -0500Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackI still see a compelling business case for a foundation supported free open source set of libs for the base protocols, commercially supported by multiple contributors who build higher level applications using the base protocols.  In a situation like this, commercial sponsors can pool resources for certification.  Obviously anyone using a modified version would not be using certified code but the sponsors who pay for certification would still get multiple benefits.MikeOn Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
I don?t see a non Commercial one getting certification and then getting widely used here in the states... That?s the problem you have to spend tons of time to get them certified... Its not like SIP where they will pretty much take anything and everything unfortunatelyKFrom: Brian West <brian at freeswitch.org>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Tue, 27 Jan 2009 12:59:18 -0600To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackWe should have an open source one too thats MPL or BSD ... I don't agree with selling protocols like this commercially.  Guess thats just the Open Source in me wanting that stuff to be free./bOn Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
There are things available for this in the commercial space.  One from CometSig.com and one from sangoma.  Nothing at this time in the opensource arena. (OpenSS7 will not be integrated with freeswitch due to licensing issues)Contact me off list if you are interested in either of the commercial versions I can assist you with these things.Ken

check out the rest of the Windows Live?. More than mail?Windows Live? goes way beyond your inbox. More than messages <http://www.microsoft.com/windows/windowslive/> 

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From shaneb at metrostat.net  Wed Jan 28 11:49:38 2009
From: shaneb at metrostat.net (Shane Burrell)
Date: Wed, 28 Jan 2009 14:49:38 -0500
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
References: <BAY125-W27F7B5DA0459D16F097397D2C80@phx.gbl>
	<C5A60FC2.4CD8A%krice@freeswitch.org>
	<BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
Message-ID: <F20767C9-528B-4A54-A19F-8625C5AE6F7F@metrostat.net>

Trust me AT&T doesn't have free docs. There are hundreds of ANSI SS7  
docs and they put alot of work and money to make sure you can't get  
them via google search.   Much of it is not in ITU but many ITU switch  
support ANSI features such as Class 5.  Even with the ITU docs in hand  
you'll need a large team to get it complete enough for Class 4/5  
operations anytime soon.
On Jan 28, 2009, at 2:42 PM, Jan Berger wrote:

> Next time try to search if AT&T have a free version of the doc first  
> - you might get a suprice.
>
>
> Date: Wed, 28 Jan 2009 13:36:02 -0600
> From: krice at freeswitch.org
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> Must be nice ;)
>
> Here in the states they want you to pay for documentation on cable  
> pinouts and don?t even think about getting documentation for  
> something like TCAP LIDB without giving up your first born and an arm.
>
>
>
>
> From: Jan Berger <janvb at live.com>
> Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org 
> >
> Date: Wed, 28 Jan 2009 20:22:38 +0100
> To: <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> I live in Europe - and ITU/ETSI are free :)
>
>
>
>
> Date: Wed, 28 Jan 2009 12:37:48 -0600
> From: krice at freeswitch.org
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> That?s the same thing we saw...  Everyone wants SS7 but no one wants  
> to help fund it... And to make matters worse just getting proper  
> documentation for ANSI SS7 is not exactly a cheap endeavor.  
> Unfortunately telecordia and crew still want to charge thousands of  
> dollars for the docs when the ITU decided ages ago to make those  
> available for free... Even then its still a substantial amount of  
> effort to get something working (even if you have active links you  
> can use to help understand whats going on)
>
> K
>
>
>
>
>
>
> From: Jan Berger <janvb at live.com>
> Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org 
> >
> Date: Wed, 28 Jan 2009 19:30:15 +0100
> To: <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> This is a little like the hen and the egg...
>
> No one will support you starting to write a SS7 ++ stack, but once  
> you get going and the first stacks turns up, so will business.
>
> Jan
>
>
>
>
>
>
> From: mike at jerris.com
> To: freeswitch-dev at lists.freeswitch.org
> Date: Tue, 27 Jan 2009 15:06:11 -0500
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
>
> I still see a compelling business case for a foundation supported  
> free open source set of libs for the base protocols, commercially  
> supported by multiple contributors who build higher level  
> applications using the base protocols.  In a situation like this,  
> commercial sponsors can pool resources for certification.  Obviously  
> anyone using a modified version would not be using certified code  
> but the sponsors who pay for certification would still get multiple  
> benefits.
>
> Mike
>
> On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
>
>
>
> I don?t see a non Commercial one getting certification and then  
> getting widely used here in the states... That?s the problem you  
> have to spend tons of time to get them certified... Its not like SIP  
> where they will pretty much take anything and everything unfortunately
>
> K
>
> From: Brian West <brian at freeswitch.org>
> Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org 
> >
> Date: Tue, 27 Jan 2009 12:59:18 -0600
> To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org 
> >
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> We should have an open source one too thats MPL or BSD ... I don't  
> agree with selling protocols like this commercially.  Guess thats  
> just the Open Source in me wanting that stuff to be free.
>
> /b
>
> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
>
> There are things available for this in the commercial space.  One  
> from CometSig.com and one from sangoma.  Nothing at this time in the  
> opensource arena. (OpenSS7 will not be integrated with freeswitch  
> due to licensing issues)
>
> Contact me off list if you are interested in either of the  
> commercial versions I can assist you with these things.
>
> Ken
>
>
> check out the rest of the Windows Live?. More than mail?Windows  
> Live? goes way beyond your inbox. More than messages <http://www.microsoft.com/windows/windowslive/ 
> >
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/>
>
> check out the rest of the Windows Live?. More than mail?Windows  
> Live? goes way beyond your inbox. More than messages <http://www.microsoft.com/windows/windowslive/ 
> >
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
> Get news, entertainment and everything you care about at Live.com.  
> Check it out! _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
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From krice at freeswitch.org  Wed Jan 28 12:02:38 2009
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 28 Jan 2009 14:02:38 -0600
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <BAY119-W3ABD90DFEA06701EE0292D2C80@phx.gbl>
Message-ID: <C5A615FE.4CD98%krice@freeswitch.org>

No SS7 Box is Mike Mueller and Sangoma,  We is Shane Burrell and I with a
box called APSS7 that does a stack of TCAP  and ALC functions along with
SIGTRAN and Traditional TDM links



From: Jan Berger <janvb at live.com>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Wed, 28 Jan 2009 20:53:52 +0100
To: <freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

"We" is that SS7box ?


Date: Wed, 28 Jan 2009 13:45:07 -0600
From: krice at freeswitch.org
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

We did haha... Searched high and low... Ended up reverse engineering part of
the spec while we tried to come up with documentation...



From: Jan Berger <janvb at live.com>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Wed, 28 Jan 2009 20:42:07 +0100
To: <freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

Next time try to search if AT&T have a free version of the doc first - you
might get a suprice.



Date: Wed, 28 Jan 2009 13:36:02 -0600
From: krice at freeswitch.org
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

Must be nice ;) 

Here in the states they want you to pay for documentation on cable pinouts
and don?t even think about getting documentation for something like TCAP
LIDB without giving up your first born and an arm.





From: Jan Berger <janvb at live.com>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Wed, 28 Jan 2009 20:22:38 +0100
To: <freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

I live in Europe - and ITU/ETSI are free :)
 




Date: Wed, 28 Jan 2009 12:37:48 -0600
From: krice at freeswitch.org
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

That?s the same thing we saw...  Everyone wants SS7 but no one wants to help
fund it... And to make matters worse just getting proper documentation for
ANSI SS7 is not exactly a cheap endeavor. Unfortunately telecordia and crew
still want to charge thousands of dollars for the docs when the ITU decided
ages ago to make those available for free... Even then its still a
substantial amount of effort to get something working (even if you have
active links you can use to help understand whats going on)

K







From: Jan Berger <janvb at live.com>
Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
Date: Wed, 28 Jan 2009 19:30:15 +0100
To: <freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

This is a little like the hen and the egg...
 
No one will support you starting to write a SS7 ++ stack, but once you get
going and the first stacks turns up, so will business.
 
Jan







From: mike at jerris.com
To: freeswitch-dev at lists.freeswitch.org
Date: Tue, 27 Jan 2009 15:06:11 -0500
Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack


I still see a compelling business case for a foundation supported free open
source set of libs for the base protocols, commercially supported by
multiple contributors who build higher level applications using the base
protocols.  In a situation like this, commercial sponsors can pool resources
for certification.  Obviously anyone using a modified version would not be
using certified code but the sponsors who pay for certification would still
get multiple benefits.

Mike

On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:



> I don?t see a non Commercial one getting certification and then getting widely
> used here in the states... That?s the problem you have to spend tons of time
> to get them certified... Its not like SIP where they will pretty much take
> anything and everything unfortunately
> 
> K
> 
> From: Brian West <brian at freeswitch.org>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Tue, 27 Jan 2009 12:59:18 -0600
> To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> 
> We should have an open source one too thats MPL or BSD ... I don't agree with
> selling protocols like this commercially.  Guess thats just the Open Source in
> me wanting that stuff to be free.
> 
> /b
> 
> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
> 
>> There are things available for this in the commercial space.  One from
>> CometSig.com and one from sangoma.  Nothing at this time in the opensource
>> arena. (OpenSS7 will not be integrated with freeswitch due to licensing
>> issues)
>> 
>> Contact me off list if you are interested in either of the commercial
>> versions I can assist you with these things.
>> 
>> Ken
> 
> 
> 
> check out the rest of the Windows Live?. More than mail?Windows Live? goes way
> beyond your inbox. More than messages
> <http://www.microsoft.com/windows/windowslive/>
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/>
> <http://www.freeswitch.org/>  <http://www.freeswitch.org/>
> 
> 
> check out the rest of the Windows Live?. More than mail?Windows Live? goes way
> beyond your inbox. More than messages
> <http://www.microsoft.com/windows/windowslive/>
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/>
> <http://www.freeswitch.org/>
> 
> 
> Get news, entertainment and everything you care about at Live.com. Check it
> out! <http://www.live.com/getstarted.aspx >
> <http://www.live.com/getstarted.aspx>
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> check out the rest of the Windows Live?. More than mail?Windows Live? goes way
> beyond your inbox. More than messages
> <http://www.microsoft.com/windows/windowslive/>
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
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From janvb at live.com  Wed Jan 28 12:07:23 2009
From: janvb at live.com (Jan Berger)
Date: Wed, 28 Jan 2009 21:07:23 +0100
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <F20767C9-528B-4A54-A19F-8625C5AE6F7F@metrostat.net>
References: <BAY125-W27F7B5DA0459D16F097397D2C80@phx.gbl>
	<C5A60FC2.4CD8A%krice@freeswitch.org>
	<BAY125-W42F46B61304888CE624147D2C80@phx.gbl> 
	<F20767C9-528B-4A54-A19F-8625C5AE6F7F@metrostat.net>
Message-ID: <BAY118-W429C55F77703B7E0F7E09CD2C80@phx.gbl>


They have docs you can download form their site - that's where I got the AT&T ISDN stacks etc 4ESS /5ESS + some BRI stuff. But, I don't know about other stacks.

From: shaneb at metrostat.netTo: freeswitch-dev at lists.freeswitch.orgDate: Wed, 28 Jan 2009 14:49:38 -0500Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackTrust me AT&T doesn't have free docs. There are hundreds of ANSI SS7 docs and they put alot of work and money to make sure you can't get them via google search.   Much of it is not in ITU but many ITU switch support ANSI features such as Class 5.  Even with the ITU docs in hand you'll need a large team to get it complete enough for Class 4/5 operations anytime soon.

On Jan 28, 2009, at 2:42 PM, Jan Berger wrote:

Next time try to search if AT&T have a free version of the doc first - you might get a suprice.

Date: Wed, 28 Jan 2009 13:36:02 -0600From: krice at freeswitch.orgTo: freeswitch-dev at lists.freeswitch.orgSubject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackMust be nice ;) Here in the states they want you to pay for documentation on cable pinouts and don?t even think about getting documentation for something like TCAP LIDB without giving up your first born and an arm.

From: Jan Berger <janvb at live.com>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Wed, 28 Jan 2009 20:22:38 +0100To: <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackI live in Europe - and ITU/ETSI are free :) 

Date: Wed, 28 Jan 2009 12:37:48 -0600From: krice at freeswitch.orgTo: freeswitch-dev at lists.freeswitch.orgSubject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackThat?s the same thing we saw...  Everyone wants SS7 but no one wants to help fund it... And to make matters worse just getting proper documentation for ANSI SS7 is not exactly a cheap endeavor. Unfortunately telecordia and crew still want to charge thousands of dollars for the docs when the ITU decided ages ago to make those available for free... Even then its still a substantial amount of effort to get something working (even if you have active links you can use to help understand whats going on)K

From: Jan Berger <janvb at live.com>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Wed, 28 Jan 2009 19:30:15 +0100To: <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackThis is a little like the hen and the egg... No one will support you starting to write a SS7 ++ stack, but once you get going and the first stacks turns up, so will business. Jan

From: mike at jerris.comTo: freeswitch-dev at lists.freeswitch.orgDate: Tue, 27 Jan 2009 15:06:11 -0500Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackI still see a compelling business case for a foundation supported free open source set of libs for the base protocols, commercially supported by multiple contributors who build higher level applications using the base protocols.  In a situation like this, commercial sponsors can pool resources for certification.  Obviously anyone using a modified version would not be using certified code but the sponsors who pay for certification would still get multiple benefits.MikeOn Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
I don?t see a non Commercial one getting certification and then getting widely used here in the states... That?s the problem you have to spend tons of time to get them certified... Its not like SIP where they will pretty much take anything and everything unfortunatelyKFrom: Brian West <brian at freeswitch.org>Reply-To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Date: Tue, 27 Jan 2009 12:59:18 -0600To: "freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stackWe should have an open source one too thats MPL or BSD ... I don't agree with selling protocols like this commercially.  Guess thats just the Open Source in me wanting that stuff to be free./bOn Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
There are things available for this in the commercial space.  One from CometSig.com and one from sangoma.  Nothing at this time in the opensource arena. (OpenSS7 will not be integrated with freeswitch due to licensing issues)Contact me off list if you are interested in either of the commercial versions I can assist you with these things.Ken

check out the rest of the Windows Live?. More than mail?Windows Live? goes way beyond your inbox. More than messages <http://www.microsoft.com/windows/windowslive/> 

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check out the rest of the Windows Live?. More than mail?Windows Live? goes way beyond your inbox. More than messages <http://www.microsoft.com/windows/windowslive/> 

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From anthony.minessale at gmail.com  Wed Jan 28 12:13:20 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Jan 2009 14:13:20 -0600
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <C5A615FE.4CD98%krice@freeswitch.org>
References: <BAY119-W3ABD90DFEA06701EE0292D2C80@phx.gbl>
	<C5A615FE.4CD98%krice@freeswitch.org>
Message-ID: <191c3a030901281213o7376f0c1x522922bdbaf19da1@mail.gmail.com>

Some people like to donate code to open source.

Thanks again Jan for the ISDN code.  We are still working on it and some day
we should make it to SS7 but it will probably take a bit more work since we
are already short on hands for people who want to work on OpenZAP
abstraction layer.



On Wed, Jan 28, 2009 at 2:02 PM, Ken Rice <krice at freeswitch.org> wrote:

>  No SS7 Box is Mike Mueller and Sangoma,  We is Shane Burrell and I with a
> box called APSS7 that does a stack of TCAP  and ALC functions along with
> SIGTRAN and Traditional TDM links
>
>
> ------------------------------
> *From: *Jan Berger <janvb at live.com>
> *Reply-To: *"freeswitch-dev at lists.freeswitch.org" <
> freeswitch-dev at lists.freeswitch.org>
> *Date: *Wed, 28 Jan 2009 20:53:52 +0100
>
> *To: *<freeswitch-dev at lists.freeswitch.org>
> *Subject: *Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> "We" is that SS7box ?
>
> ------------------------------
> Date: Wed, 28 Jan 2009 13:45:07 -0600
> From: krice at freeswitch.org
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> We did haha... Searched high and low... Ended up reverse engineering part
> of the spec while we tried to come up with documentation...
>
>
> ------------------------------
> *From: *Jan Berger <janvb at live.com>
> *Reply-To: *"freeswitch-dev at lists.freeswitch.org" <
> freeswitch-dev at lists.freeswitch.org>
> *Date: *Wed, 28 Jan 2009 20:42:07 +0100
> *To: *<freeswitch-dev at lists.freeswitch.org>
> *Subject: *Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> Next time try to search if AT&T have a free version of the doc first - you
> might get a suprice.
>
> ------------------------------
>
> Date: Wed, 28 Jan 2009 13:36:02 -0600
> From: krice at freeswitch.org
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> Must be nice ;)
>
> Here in the states they want you to pay for documentation on cable pinouts
> and don't even think about getting documentation for something like TCAP
> LIDB without giving up your first born and an arm.
>
>
>
> ------------------------------
>
> *From: *Jan Berger <janvb at live.com>
> *Reply-To: *"freeswitch-dev at lists.freeswitch.org" <
> freeswitch-dev at lists.freeswitch.org>
> *Date: *Wed, 28 Jan 2009 20:22:38 +0100
> *To: *<freeswitch-dev at lists.freeswitch.org>
> *Subject: *Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> I live in Europe - and ITU/ETSI are free :)
>
>
> ------------------------------
>
>
> Date: Wed, 28 Jan 2009 12:37:48 -0600
> From: krice at freeswitch.org
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> That's the same thing we saw...  Everyone wants SS7 but no one wants to
> help fund it... And to make matters worse just getting proper documentation
> for ANSI SS7 is not exactly a cheap endeavor. Unfortunately telecordia and
> crew still want to charge thousands of dollars for the docs when the ITU
> decided ages ago to make those available for free... Even then its still a
> substantial amount of effort to get something working (even if you have
> active links you can use to help understand whats going on)
>
> K
>
>
>
>
> ------------------------------
>
>
> *From: *Jan Berger <janvb at live.com>
> *Reply-To: *"freeswitch-dev at lists.freeswitch.org" <
> freeswitch-dev at lists.freeswitch.org>
> *Date: *Wed, 28 Jan 2009 19:30:15 +0100
> *To: *<freeswitch-dev at lists.freeswitch.org>
> *Subject: *Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> This is a little like the hen and the egg...
>
> No one will support you starting to write a SS7 ++ stack, but once you get
> going and the first stacks turns up, so will business.
>
> Jan
>
>
>
> ------------------------------
>
>
>
> From: mike at jerris.com
> To: freeswitch-dev at lists.freeswitch.org
> Date: Tue, 27 Jan 2009 15:06:11 -0500
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
>
> I still see a compelling business case for a foundation supported free open
> source set of libs for the base protocols, commercially supported by
> multiple contributors who build higher level applications using the base
> protocols.  In a situation like this, commercial sponsors can pool resources
> for certification.  Obviously anyone using a modified version would not be
> using certified code but the sponsors who pay for certification would still
> get multiple benefits.
>
> Mike
>
> On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
>
>
>
> I don't see a non Commercial one getting certification and then getting
> widely used here in the states... That's the problem you have to spend tons
> of time to get them certified... Its not like SIP where they will pretty
> much take anything and everything unfortunately
>
> K
>
> *From: *Brian West <brian at freeswitch.org>
> *Reply-To: *"freeswitch-dev at lists.freeswitch.org" <
> freeswitch-dev at lists.freeswitch.org>
> *Date: *Tue, 27 Jan 2009 12:59:18 -0600
> *To: *"freeswitch-dev at lists.freeswitch.org" <
> freeswitch-dev at lists.freeswitch.org>
> *Subject: *Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> We should have an open source one too thats MPL or BSD ... I don't agree
> with selling protocols like this commercially.  Guess thats just the Open
> Source in me wanting that stuff to be free.
>
> /b
>
> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
>
> There are things available for this in the commercial space.  One from
> CometSig.com and one from sangoma.  Nothing at this time in the opensource
> arena. (OpenSS7 will not be integrated with freeswitch due to licensing
> issues)
>
> Contact me off list if you are interested in either of the commercial
> versions I can assist you with these things.
>
> Ken
>
>
>
> ------------------------------
> check out the rest of the Windows Live?. More than mail?Windows Live? goes
> way beyond your inbox. More than messages
> <http://www.microsoft.com/windows/windowslive/><http://www.microsoft.com/windows/windowslive/>
> ------------------------------
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/><http://www.freeswitch.org/>
> <http://www.freeswitch.org/> <http://www.freeswitch.org/>
> <http://www.freeswitch.org/> <http://www.freeswitch.org/>
>
> ------------------------------
> check out the rest of the Windows Live?. More than mail?Windows Live? goes
> way beyond your inbox. More than messages
> <http://www.microsoft.com/windows/windowslive/><http://www.microsoft.com/windows/windowslive/>
> ------------------------------
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/><http://www.freeswitch.org/>
> <http://www.freeswitch.org/> <http://www.freeswitch.org/>
>
> ------------------------------
> Get news, entertainment and everything you care about at Live.com. Check it
> out! <http://www.live.com/getstarted.aspx ><http://www.live.com/getstarted.aspx>
> <http://www.live.com/getstarted.aspx><http://www.live.com/getstarted.aspx>
> ------------------------------
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/><http://www.freeswitch.org/>
>
> ------------------------------
> check out the rest of the Windows Live?. More than mail?Windows Live? goes
> way beyond your inbox. More than messages
> <http://www.microsoft.com/windows/windowslive/><http://www.microsoft.com/windows/windowslive/>
> ------------------------------
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From R.Kloosterman at mtel.nl  Wed Jan 28 12:47:16 2009
From: R.Kloosterman at mtel.nl (Remko Kloosterman)
Date: Wed, 28 Jan 2009 21:47:16 +0100
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <F20767C9-528B-4A54-A19F-8625C5AE6F7F@metrostat.net>
References: <BAY125-W27F7B5DA0459D16F097397D2C80@phx.gbl>
	<C5A60FC2.4CD8A%krice@freeswitch.org>
	<BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<F20767C9-528B-4A54-A19F-8625C5AE6F7F@metrostat.net>
Message-ID: <11372C8B9E603F4FACDE6AB18256DEC695A931@srvmtel.office.mtel.nl>

I've done some SS7 development in the past, using dedicated TX3220 and
TX4000 signalling boards from NMS Communications (recentlty aquired by
Dialogic). Even with a lot of commercial documentation available, it's
horribly complex stuff, especially redundancy. And you need another SS7
switch to test against for interoperability. And SIP trunks are becoming
more and more commonly available these days, so one can question if it's
worth the effort. Still, I would really like it if someone would try and
develop such a feature.
 
Off-topic: If you need SS7 and are willing to pay for it, I can
recommend audiocodes mediant gateways. There's a model available with
embedded SS7 signalling, provided by Teles. And real cheap too, compared
to other SS7 switches.
 

________________________________

Van: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] Namens Shane
Burrell
Verzonden: woensdag 28 januari 2009 20:50
Aan: freeswitch-dev at lists.freeswitch.org
Onderwerp: Re: [Freeswitch-dev] FreeSWITCH SS7 stack


Trust me AT&T doesn't have free docs. There are hundreds of ANSI SS7
docs and they put alot of work and money to make sure you can't get them
via google search.   Much of it is not in ITU but many ITU switch
support ANSI features such as Class 5.  Even with the ITU docs in hand
you'll need a large team to get it complete enough for Class 4/5
operations anytime soon.

On Jan 28, 2009, at 2:42 PM, Jan Berger wrote:


	
	Next time try to search if AT&T have a free version of the doc
first - you might get a suprice.
	
	
________________________________


	Date: Wed, 28 Jan 2009 13:36:02 -0600
	From: krice at freeswitch.org
	To: freeswitch-dev at lists.freeswitch.org
	Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
	
	Must be nice ;) 
	
	Here in the states they want you to pay for documentation on
cable pinouts and don't even think about getting documentation for
something like TCAP LIDB without giving up your first born and an arm.
	
	
	
	
________________________________


	From: Jan Berger <janvb at live.com>
	Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
	Date: Wed, 28 Jan 2009 20:22:38 +0100
	To: <freeswitch-dev at lists.freeswitch.org>
	Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
	
	I live in Europe - and ITU/ETSI are free :)
	 
	
	
________________________________



	Date: Wed, 28 Jan 2009 12:37:48 -0600
	From: krice at freeswitch.org
	To: freeswitch-dev at lists.freeswitch.org
	Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
	
	That's the same thing we saw...  Everyone wants SS7 but no one
wants to help fund it... And to make matters worse just getting proper
documentation for ANSI SS7 is not exactly a cheap endeavor.
Unfortunately telecordia and crew still want to charge thousands of
dollars for the docs when the ITU decided ages ago to make those
available for free... Even then its still a substantial amount of effort
to get something working (even if you have active links you can use to
help understand whats going on)
	
	K
	
	
	
	
	
________________________________



	From: Jan Berger <janvb at live.com>
	Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
	Date: Wed, 28 Jan 2009 19:30:15 +0100
	To: <freeswitch-dev at lists.freeswitch.org>
	Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
	
	This is a little like the hen and the egg...
	 
	No one will support you starting to write a SS7 ++ stack, but
once you get going and the first stacks turns up, so will business.
	 
	Jan
	
	
	
	
________________________________




	From: mike at jerris.com
	To: freeswitch-dev at lists.freeswitch.org
	Date: Tue, 27 Jan 2009 15:06:11 -0500
	Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
	
	
	I still see a compelling business case for a foundation
supported free open source set of libs for the base protocols,
commercially supported by multiple contributors who build higher level
applications using the base protocols.  In a situation like this,
commercial sponsors can pool resources for certification.  Obviously
anyone using a modified version would not be using certified code but
the sponsors who pay for certification would still get multiple
benefits.
	
	Mike
	
	On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
	
	
	
	

		I don't see a non Commercial one getting certification
and then getting widely used here in the states... That's the problem
you have to spend tons of time to get them certified... Its not like SIP
where they will pretty much take anything and everything unfortunately
		
		K
		
		From: Brian West <brian at freeswitch.org>
		Reply-To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
		Date: Tue, 27 Jan 2009 12:59:18 -0600
		To: "freeswitch-dev at lists.freeswitch.org"
<freeswitch-dev at lists.freeswitch.org>
		Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
		
		We should have an open source one too thats MPL or BSD
... I don't agree with selling protocols like this commercially.  Guess
thats just the Open Source in me wanting that stuff to be free.
		
		/b
		
		On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
		
		

			There are things available for this in the
commercial space.  One from CometSig.com and one from sangoma.  Nothing
at this time in the opensource arena. (OpenSS7 will not be integrated
with freeswitch due to licensing issues)
			
			Contact me off list if you are interested in
either of the commercial versions I can assist you with these things.
			
			Ken
			

		
		
		
________________________________

		check out the rest of the Windows Live(tm). More than
mail-Windows Live(tm) goes way beyond your inbox. More than messages
<http://www.microsoft.com/windows/windowslive/>
<http://www.microsoft.com/windows/windowslive/>  
		
________________________________

		_______________________________________________
		Freeswitch-dev mailing list
		Freeswitch-dev at lists.freeswitch.org
	
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
	
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
		http://www.freeswitch.org <http://www.freeswitch.org/>
<http://www.freeswitch.org/> <http://www.freeswitch.org/>  
		
		
________________________________

		check out the rest of the Windows Live(tm). More than
mail-Windows Live(tm) goes way beyond your inbox. More than messages
<http://www.microsoft.com/windows/windowslive/>
<http://www.microsoft.com/windows/windowslive/>  
		
________________________________


	_______________________________________________
	Freeswitch-dev mailing list
	Freeswitch-dev at lists.freeswitch.org
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
	
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
	http://www.freeswitch.org <http://www.freeswitch.org/> 
	
	
________________________________

	Get news, entertainment and everything you care about at
Live.com. Check it out! <http://www.live.com/getstarted.aspx>
_______________________________________________
	Freeswitch-dev mailing list
	Freeswitch-dev at lists.freeswitch.org
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
	
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
	http://www.freeswitch.org
	


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From msc at freeswitch.org  Wed Jan 28 13:23:12 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Jan 2009 13:23:12 -0800
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC695A931@srvmtel.office.mtel.nl>
References: <BAY125-W27F7B5DA0459D16F097397D2C80@phx.gbl>
	<C5A60FC2.4CD8A%krice@freeswitch.org>
	<BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<F20767C9-528B-4A54-A19F-8625C5AE6F7F@metrostat.net>
	<11372C8B9E603F4FACDE6AB18256DEC695A931@srvmtel.office.mtel.nl>
Message-ID: <87f2f3b90901281323n28f4076eyffdfb961986eacf8@mail.gmail.com>

How about some SS7 documentation? ;)
-MC

On Wed, Jan 28, 2009 at 12:47 PM, Remko Kloosterman
<R.Kloosterman at mtel.nl> wrote:
> I've done some SS7 development in the past, using dedicated TX3220 and
> TX4000 signalling boards from NMS Communications (recentlty aquired by
> Dialogic). Even with a lot of commercial documentation available, it's
> horribly complex stuff, especially redundancy. And you need another SS7
> switch to test against for interoperability. And SIP trunks are becoming
> more and more commonly available these days, so one can question if it's
> worth the effort. Still, I would really like it if someone would try and
> develop such a feature.
>
> Off-topic: If you need SS7 and are willing to pay for it, I can recommend
> audiocodes mediant gateways. There's a model available with embedded SS7
> signalling, provided by Teles. And real cheap too, compared to other SS7
> switches.
>
> ________________________________
> Van: freeswitch-dev-bounces at lists.freeswitch.org
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Namens Shane Burrell
> Verzonden: woensdag 28 januari 2009 20:50
> Aan: freeswitch-dev at lists.freeswitch.org
> Onderwerp: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> Trust me AT&T doesn't have free docs. There are hundreds of ANSI SS7 docs
> and they put alot of work and money to make sure you can't get them via
> google search.   Much of it is not in ITU but many ITU switch support ANSI
> features such as Class 5.  Even with the ITU docs in hand you'll need a
> large team to get it complete enough for Class 4/5 operations anytime soon.
> On Jan 28, 2009, at 2:42 PM, Jan Berger wrote:
>
> Next time try to search if AT&T have a free version of the doc first - you
> might get a suprice.
>
> ________________________________
> Date: Wed, 28 Jan 2009 13:36:02 -0600
> From: krice at freeswitch.org
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> Must be nice ;)
>
> Here in the states they want you to pay for documentation on cable pinouts
> and don't even think about getting documentation for something like TCAP
> LIDB without giving up your first born and an arm.
>
>
>
> ________________________________
> From: Jan Berger <janvb at live.com>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Wed, 28 Jan 2009 20:22:38 +0100
> To: <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> I live in Europe - and ITU/ETSI are free :)
>
>
> ________________________________
>
> Date: Wed, 28 Jan 2009 12:37:48 -0600
> From: krice at freeswitch.org
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> That's the same thing we saw...  Everyone wants SS7 but no one wants to help
> fund it... And to make matters worse just getting proper documentation for
> ANSI SS7 is not exactly a cheap endeavor. Unfortunately telecordia and crew
> still want to charge thousands of dollars for the docs when the ITU decided
> ages ago to make those available for free... Even then its still a
> substantial amount of effort to get something working (even if you have
> active links you can use to help understand whats going on)
>
> K
>
>
>
>
> ________________________________
>
> From: Jan Berger <janvb at live.com>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Wed, 28 Jan 2009 19:30:15 +0100
> To: <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> This is a little like the hen and the egg...
>
> No one will support you starting to write a SS7 ++ stack, but once you get
> going and the first stacks turns up, so will business.
>
> Jan
>
>
>
> ________________________________
>
>
> From: mike at jerris.com
> To: freeswitch-dev at lists.freeswitch.org
> Date: Tue, 27 Jan 2009 15:06:11 -0500
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
>
> I still see a compelling business case for a foundation supported free open
> source set of libs for the base protocols, commercially supported by
> multiple contributors who build higher level applications using the base
> protocols.  In a situation like this, commercial sponsors can pool resources
> for certification.  Obviously anyone using a modified version would not be
> using certified code but the sponsors who pay for certification would still
> get multiple benefits.
>
> Mike
>
> On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
>
>
>
> I don't see a non Commercial one getting certification and then getting
> widely used here in the states... That's the problem you have to spend tons
> of time to get them certified... Its not like SIP where they will pretty
> much take anything and everything unfortunately
>
> K
>
> From: Brian West <brian at freeswitch.org>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Tue, 27 Jan 2009 12:59:18 -0600
> To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> We should have an open source one too thats MPL or BSD ... I don't agree
> with selling protocols like this commercially.  Guess thats just the Open
> Source in me wanting that stuff to be free.
>
> /b
>
> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
>
> There are things available for this in the commercial space.  One from
> CometSig.com and one from sangoma.  Nothing at this time in the opensource
> arena. (OpenSS7 will not be integrated with freeswitch due to licensing
> issues)
>
> Contact me off list if you are interested in either of the commercial
> versions I can assist you with these things.
>
> Ken
>
>
> ________________________________
> check out the rest of the Windows Live?. More than mail?Windows Live? goes
> way beyond your inbox. More than
> messages <http://www.microsoft.com/windows/windowslive/>
> ________________________________
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/>
>
> ________________________________
> check out the rest of the Windows Live?. More than mail?Windows Live? goes
> way beyond your inbox. More than
> messages <http://www.microsoft.com/windows/windowslive/>
> ________________________________
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
> ________________________________
> Get news, entertainment and everything you care about at Live.com. Check it
> out! _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


From R.Kloosterman at mtel.nl  Wed Jan 28 15:01:25 2009
From: R.Kloosterman at mtel.nl (Remko Kloosterman)
Date: Thu, 29 Jan 2009 00:01:25 +0100
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <87f2f3b90901281323n28f4076eyffdfb961986eacf8@mail.gmail.com>
References: <BAY125-W27F7B5DA0459D16F097397D2C80@phx.gbl>
	<C5A60FC2.4CD8A%krice@freeswitch.org>
	<BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<F20767C9-528B-4A54-A19F-8625C5AE6F7F@metrostat.net>
	<11372C8B9E603F4FACDE6AB18256DEC695A931@srvmtel.office.mtel.nl>
	<87f2f3b90901281323n28f4076eyffdfb961986eacf8@mail.gmail.com>
Message-ID: <11372C8B9E603F4FACDE6AB18256DEC695A933@srvmtel.office.mtel.nl>

Well, I cannot disclose the commercial stuff, but I can share some
experience and resources that I've found to be useful. For example:

Getting started: http://pt.com/page/tutorials/ss7-tutorial/
ITU recommendations: http://eu.sabotage.org/, dig in to www/ITU.
I guess you know www.openss7.org
Online (but recently restricted) MTP3 decoder:
http://www.linkbit.com/decoder/decoder.html
Perhaps some design inspiration:
http://www.nmscommunications.com/DevPlatforms/Support/SWandDoc/default.h
tm?section=4&ID=43&PAC=472&ST=Current&dok=

Want more?
Remko


-----Oorspronkelijk bericht-----
Van: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] Namens Michael
Collins
Verzonden: woensdag 28 januari 2009 22:23
Aan: freeswitch-dev at lists.freeswitch.org
Onderwerp: Re: [Freeswitch-dev] FreeSWITCH SS7 stack

How about some SS7 documentation? ;)
-MC

On Wed, Jan 28, 2009 at 12:47 PM, Remko Kloosterman
<R.Kloosterman at mtel.nl> wrote:
> I've done some SS7 development in the past, using dedicated TX3220 and

> TX4000 signalling boards from NMS Communications (recentlty aquired by

> Dialogic). Even with a lot of commercial documentation available, it's

> horribly complex stuff, especially redundancy. And you need another 
> SS7 switch to test against for interoperability. And SIP trunks are 
> becoming more and more commonly available these days, so one can 
> question if it's worth the effort. Still, I would really like it if 
> someone would try and develop such a feature.
>
> Off-topic: If you need SS7 and are willing to pay for it, I can 
> recommend audiocodes mediant gateways. There's a model available with 
> embedded SS7 signalling, provided by Teles. And real cheap too, 
> compared to other SS7 switches.
>
> ________________________________
> Van: freeswitch-dev-bounces at lists.freeswitch.org
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Namens Shane 
> Burrell
> Verzonden: woensdag 28 januari 2009 20:50
> Aan: freeswitch-dev at lists.freeswitch.org
> Onderwerp: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> Trust me AT&T doesn't have free docs. There are hundreds of ANSI SS7 
> docs and they put alot of work and money to make sure you can't get
them via
> google search.   Much of it is not in ITU but many ITU switch support
ANSI
> features such as Class 5.  Even with the ITU docs in hand you'll need 
> a large team to get it complete enough for Class 4/5 operations
anytime soon.
> On Jan 28, 2009, at 2:42 PM, Jan Berger wrote:
>
> Next time try to search if AT&T have a free version of the doc first -

> you might get a suprice.
>
> ________________________________
> Date: Wed, 28 Jan 2009 13:36:02 -0600
> From: krice at freeswitch.org
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> Must be nice ;)
>
> Here in the states they want you to pay for documentation on cable 
> pinouts and don't even think about getting documentation for something

> like TCAP LIDB without giving up your first born and an arm.
>
>
>
> ________________________________
> From: Jan Berger <janvb at live.com>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Wed, 28 Jan 2009 20:22:38 +0100
> To: <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> I live in Europe - and ITU/ETSI are free :)
>
>
> ________________________________
>
> Date: Wed, 28 Jan 2009 12:37:48 -0600
> From: krice at freeswitch.org
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> That's the same thing we saw...  Everyone wants SS7 but no one wants 
> to help fund it... And to make matters worse just getting proper 
> documentation for ANSI SS7 is not exactly a cheap endeavor. 
> Unfortunately telecordia and crew still want to charge thousands of 
> dollars for the docs when the ITU decided ages ago to make those 
> available for free... Even then its still a substantial amount of 
> effort to get something working (even if you have active links you can

> use to help understand whats going on)
>
> K
>
>
>
>
> ________________________________
>
> From: Jan Berger <janvb at live.com>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Wed, 28 Jan 2009 19:30:15 +0100
> To: <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> This is a little like the hen and the egg...
>
> No one will support you starting to write a SS7 ++ stack, but once you

> get going and the first stacks turns up, so will business.
>
> Jan
>
>
>
> ________________________________
>
>
> From: mike at jerris.com
> To: freeswitch-dev at lists.freeswitch.org
> Date: Tue, 27 Jan 2009 15:06:11 -0500
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
>
> I still see a compelling business case for a foundation supported free

> open source set of libs for the base protocols, commercially supported

> by multiple contributors who build higher level applications using the

> base protocols.  In a situation like this, commercial sponsors can 
> pool resources for certification.  Obviously anyone using a modified 
> version would not be using certified code but the sponsors who pay for

> certification would still get multiple benefits.
>
> Mike
>
> On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
>
>
>
> I don't see a non Commercial one getting certification and then 
> getting widely used here in the states... That's the problem you have 
> to spend tons of time to get them certified... Its not like SIP where 
> they will pretty much take anything and everything unfortunately
>
> K
>
> From: Brian West <brian at freeswitch.org>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Tue, 27 Jan 2009 12:59:18 -0600
> To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> We should have an open source one too thats MPL or BSD ... I don't 
> agree with selling protocols like this commercially.  Guess thats just

> the Open Source in me wanting that stuff to be free.
>
> /b
>
> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
>
> There are things available for this in the commercial space.  One from

> CometSig.com and one from sangoma.  Nothing at this time in the 
> opensource arena. (OpenSS7 will not be integrated with freeswitch due 
> to licensing
> issues)
>
> Contact me off list if you are interested in either of the commercial 
> versions I can assist you with these things.
>
> Ken
>
>
> ________________________________
> check out the rest of the Windows Live(tm). More than mail-Windows
Live(tm) 
> goes way beyond your inbox. More than messages 
> <http://www.microsoft.com/windows/windowslive/>
> ________________________________
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org <http://www.freeswitch.org/>
>
> ________________________________
> check out the rest of the Windows Live(tm). More than mail-Windows
Live(tm) 
> goes way beyond your inbox. More than messages 
> <http://www.microsoft.com/windows/windowslive/>
> ________________________________
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
> ________________________________
> Get news, entertainment and everything you care about at Live.com. 
> Check it out! _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>

_______________________________________________
Freeswitch-dev mailing list
Freeswitch-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org


From msc at freeswitch.org  Wed Jan 28 15:07:39 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Jan 2009 15:07:39 -0800
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC695A933@srvmtel.office.mtel.nl>
References: <BAY125-W27F7B5DA0459D16F097397D2C80@phx.gbl>
	<C5A60FC2.4CD8A%krice@freeswitch.org>
	<BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<F20767C9-528B-4A54-A19F-8625C5AE6F7F@metrostat.net>
	<11372C8B9E603F4FACDE6AB18256DEC695A931@srvmtel.office.mtel.nl>
	<87f2f3b90901281323n28f4076eyffdfb961986eacf8@mail.gmail.com>
	<11372C8B9E603F4FACDE6AB18256DEC695A933@srvmtel.office.mtel.nl>
Message-ID: <87f2f3b90901281507v43c0c8acnb59ef6fe2b9518af@mail.gmail.com>

This is very helpful, thanks!
-MC

On Wed, Jan 28, 2009 at 3:01 PM, Remko Kloosterman
<R.Kloosterman at mtel.nl> wrote:
> Well, I cannot disclose the commercial stuff, but I can share some
> experience and resources that I've found to be useful. For example:
>
> Getting started: http://pt.com/page/tutorials/ss7-tutorial/
> ITU recommendations: http://eu.sabotage.org/, dig in to www/ITU.
> I guess you know www.openss7.org
> Online (but recently restricted) MTP3 decoder:
> http://www.linkbit.com/decoder/decoder.html
> Perhaps some design inspiration:
> http://www.nmscommunications.com/DevPlatforms/Support/SWandDoc/default.h
> tm?section=4&ID=43&PAC=472&ST=Current&dok=
>
> Want more?
> Remko
>
>
> -----Oorspronkelijk bericht-----
> Van: freeswitch-dev-bounces at lists.freeswitch.org
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Namens Michael
> Collins
> Verzonden: woensdag 28 januari 2009 22:23
> Aan: freeswitch-dev at lists.freeswitch.org
> Onderwerp: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>
> How about some SS7 documentation? ;)
> -MC
>
> On Wed, Jan 28, 2009 at 12:47 PM, Remko Kloosterman
> <R.Kloosterman at mtel.nl> wrote:
>> I've done some SS7 development in the past, using dedicated TX3220 and
>
>> TX4000 signalling boards from NMS Communications (recentlty aquired by
>
>> Dialogic). Even with a lot of commercial documentation available, it's
>
>> horribly complex stuff, especially redundancy. And you need another
>> SS7 switch to test against for interoperability. And SIP trunks are
>> becoming more and more commonly available these days, so one can
>> question if it's worth the effort. Still, I would really like it if
>> someone would try and develop such a feature.
>>
>> Off-topic: If you need SS7 and are willing to pay for it, I can
>> recommend audiocodes mediant gateways. There's a model available with
>> embedded SS7 signalling, provided by Teles. And real cheap too,
>> compared to other SS7 switches.
>>
>> ________________________________
>> Van: freeswitch-dev-bounces at lists.freeswitch.org
>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Namens Shane
>> Burrell
>> Verzonden: woensdag 28 januari 2009 20:50
>> Aan: freeswitch-dev at lists.freeswitch.org
>> Onderwerp: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>>
>> Trust me AT&T doesn't have free docs. There are hundreds of ANSI SS7
>> docs and they put alot of work and money to make sure you can't get
> them via
>> google search.   Much of it is not in ITU but many ITU switch support
> ANSI
>> features such as Class 5.  Even with the ITU docs in hand you'll need
>> a large team to get it complete enough for Class 4/5 operations
> anytime soon.
>> On Jan 28, 2009, at 2:42 PM, Jan Berger wrote:
>>
>> Next time try to search if AT&T have a free version of the doc first -
>
>> you might get a suprice.
>>
>> ________________________________
>> Date: Wed, 28 Jan 2009 13:36:02 -0600
>> From: krice at freeswitch.org
>> To: freeswitch-dev at lists.freeswitch.org
>> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>>
>> Must be nice ;)
>>
>> Here in the states they want you to pay for documentation on cable
>> pinouts and don't even think about getting documentation for something
>
>> like TCAP LIDB without giving up your first born and an arm.
>>
>>
>>
>> ________________________________
>> From: Jan Berger <janvb at live.com>
>> Reply-To: "freeswitch-dev at lists.freeswitch.org"
>> <freeswitch-dev at lists.freeswitch.org>
>> Date: Wed, 28 Jan 2009 20:22:38 +0100
>> To: <freeswitch-dev at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>>
>> I live in Europe - and ITU/ETSI are free :)
>>
>>
>> ________________________________
>>
>> Date: Wed, 28 Jan 2009 12:37:48 -0600
>> From: krice at freeswitch.org
>> To: freeswitch-dev at lists.freeswitch.org
>> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>>
>> That's the same thing we saw...  Everyone wants SS7 but no one wants
>> to help fund it... And to make matters worse just getting proper
>> documentation for ANSI SS7 is not exactly a cheap endeavor.
>> Unfortunately telecordia and crew still want to charge thousands of
>> dollars for the docs when the ITU decided ages ago to make those
>> available for free... Even then its still a substantial amount of
>> effort to get something working (even if you have active links you can
>
>> use to help understand whats going on)
>>
>> K
>>
>>
>>
>>
>> ________________________________
>>
>> From: Jan Berger <janvb at live.com>
>> Reply-To: "freeswitch-dev at lists.freeswitch.org"
>> <freeswitch-dev at lists.freeswitch.org>
>> Date: Wed, 28 Jan 2009 19:30:15 +0100
>> To: <freeswitch-dev at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>>
>> This is a little like the hen and the egg...
>>
>> No one will support you starting to write a SS7 ++ stack, but once you
>
>> get going and the first stacks turns up, so will business.
>>
>> Jan
>>
>>
>>
>> ________________________________
>>
>>
>> From: mike at jerris.com
>> To: freeswitch-dev at lists.freeswitch.org
>> Date: Tue, 27 Jan 2009 15:06:11 -0500
>> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>>
>>
>> I still see a compelling business case for a foundation supported free
>
>> open source set of libs for the base protocols, commercially supported
>
>> by multiple contributors who build higher level applications using the
>
>> base protocols.  In a situation like this, commercial sponsors can
>> pool resources for certification.  Obviously anyone using a modified
>> version would not be using certified code but the sponsors who pay for
>
>> certification would still get multiple benefits.
>>
>> Mike
>>
>> On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
>>
>>
>>
>> I don't see a non Commercial one getting certification and then
>> getting widely used here in the states... That's the problem you have
>> to spend tons of time to get them certified... Its not like SIP where
>> they will pretty much take anything and everything unfortunately
>>
>> K
>>
>> From: Brian West <brian at freeswitch.org>
>> Reply-To: "freeswitch-dev at lists.freeswitch.org"
>> <freeswitch-dev at lists.freeswitch.org>
>> Date: Tue, 27 Jan 2009 12:59:18 -0600
>> To: "freeswitch-dev at lists.freeswitch.org"
>> <freeswitch-dev at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>>
>> We should have an open source one too thats MPL or BSD ... I don't
>> agree with selling protocols like this commercially.  Guess thats just
>
>> the Open Source in me wanting that stuff to be free.
>>
>> /b
>>
>> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
>>
>> There are things available for this in the commercial space.  One from
>
>> CometSig.com and one from sangoma.  Nothing at this time in the
>> opensource arena. (OpenSS7 will not be integrated with freeswitch due
>> to licensing
>> issues)
>>
>> Contact me off list if you are interested in either of the commercial
>> versions I can assist you with these things.
>>
>> Ken
>>
>>
>> ________________________________
>> check out the rest of the Windows Live(tm). More than mail-Windows
> Live(tm)
>> goes way beyond your inbox. More than messages
>> <http://www.microsoft.com/windows/windowslive/>
>> ________________________________
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>> ________________________________
>> check out the rest of the Windows Live(tm). More than mail-Windows
> Live(tm)
>> goes way beyond your inbox. More than messages
>> <http://www.microsoft.com/windows/windowslive/>
>> ________________________________
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>> ________________________________
>> Get news, entertainment and everything you care about at Live.com.
>> Check it out! _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>


From william at channelxstream.com  Wed Jan 28 15:24:20 2009
From: william at channelxstream.com (William King)
Date: Wed, 28 Jan 2009 15:24:20 -0800
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <87f2f3b90901281507v43c0c8acnb59ef6fe2b9518af@mail.gmail.com>
References: <BAY125-W27F7B5DA0459D16F097397D2C80@phx.gbl>
	<C5A60FC2.4CD8A%krice@freeswitch.org>
	<BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<F20767C9-528B-4A54-A19F-8625C5AE6F7F@metrostat.net>
	<11372C8B9E603F4FACDE6AB18256DEC695A931@srvmtel.office.mtel.nl>
	<87f2f3b90901281323n28f4076eyffdfb961986eacf8@mail.gmail.com>
	<11372C8B9E603F4FACDE6AB18256DEC695A933@srvmtel.office.mtel.nl>
	<87f2f3b90901281507v43c0c8acnb59ef6fe2b9518af@mail.gmail.com>
Message-ID: <1233185060.22743.1.camel@quentusrex-desktop>

Way to go MikeC, taking a lively discussion on the e-mail list serve and
converting it into useful wiki docs. :)

-William 

On Wed, 2009-01-28 at 15:07 -0800, Michael Collins wrote:
> This is very helpful, thanks!
> -MC
> 
> On Wed, Jan 28, 2009 at 3:01 PM, Remko Kloosterman
> <R.Kloosterman at mtel.nl> wrote:
> > Well, I cannot disclose the commercial stuff, but I can share some
> > experience and resources that I've found to be useful. For example:
> >
> > Getting started: http://pt.com/page/tutorials/ss7-tutorial/
> > ITU recommendations: http://eu.sabotage.org/, dig in to www/ITU.
> > I guess you know www.openss7.org
> > Online (but recently restricted) MTP3 decoder:
> > http://www.linkbit.com/decoder/decoder.html
> > Perhaps some design inspiration:
> > http://www.nmscommunications.com/DevPlatforms/Support/SWandDoc/default.h
> > tm?section=4&ID=43&PAC=472&ST=Current&dok=
> >
> > Want more?
> > Remko
> >
> >
> > -----Oorspronkelijk bericht-----
> > Van: freeswitch-dev-bounces at lists.freeswitch.org
> > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Namens Michael
> > Collins
> > Verzonden: woensdag 28 januari 2009 22:23
> > Aan: freeswitch-dev at lists.freeswitch.org
> > Onderwerp: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> >
> > How about some SS7 documentation? ;)
> > -MC
> >
> > On Wed, Jan 28, 2009 at 12:47 PM, Remko Kloosterman
> > <R.Kloosterman at mtel.nl> wrote:
> >> I've done some SS7 development in the past, using dedicated TX3220 and
> >
> >> TX4000 signalling boards from NMS Communications (recentlty aquired by
> >
> >> Dialogic). Even with a lot of commercial documentation available, it's
> >
> >> horribly complex stuff, especially redundancy. And you need another
> >> SS7 switch to test against for interoperability. And SIP trunks are
> >> becoming more and more commonly available these days, so one can
> >> question if it's worth the effort. Still, I would really like it if
> >> someone would try and develop such a feature.
> >>
> >> Off-topic: If you need SS7 and are willing to pay for it, I can
> >> recommend audiocodes mediant gateways. There's a model available with
> >> embedded SS7 signalling, provided by Teles. And real cheap too,
> >> compared to other SS7 switches.
> >>
> >> ________________________________
> >> Van: freeswitch-dev-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Namens Shane
> >> Burrell
> >> Verzonden: woensdag 28 januari 2009 20:50
> >> Aan: freeswitch-dev at lists.freeswitch.org
> >> Onderwerp: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> >>
> >> Trust me AT&T doesn't have free docs. There are hundreds of ANSI SS7
> >> docs and they put alot of work and money to make sure you can't get
> > them via
> >> google search.   Much of it is not in ITU but many ITU switch support
> > ANSI
> >> features such as Class 5.  Even with the ITU docs in hand you'll need
> >> a large team to get it complete enough for Class 4/5 operations
> > anytime soon.
> >> On Jan 28, 2009, at 2:42 PM, Jan Berger wrote:
> >>
> >> Next time try to search if AT&T have a free version of the doc first -
> >
> >> you might get a suprice.
> >>
> >> ________________________________
> >> Date: Wed, 28 Jan 2009 13:36:02 -0600
> >> From: krice at freeswitch.org
> >> To: freeswitch-dev at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> >>
> >> Must be nice ;)
> >>
> >> Here in the states they want you to pay for documentation on cable
> >> pinouts and don't even think about getting documentation for something
> >
> >> like TCAP LIDB without giving up your first born and an arm.
> >>
> >>
> >>
> >> ________________________________
> >> From: Jan Berger <janvb at live.com>
> >> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> >> <freeswitch-dev at lists.freeswitch.org>
> >> Date: Wed, 28 Jan 2009 20:22:38 +0100
> >> To: <freeswitch-dev at lists.freeswitch.org>
> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> >>
> >> I live in Europe - and ITU/ETSI are free :)
> >>
> >>
> >> ________________________________
> >>
> >> Date: Wed, 28 Jan 2009 12:37:48 -0600
> >> From: krice at freeswitch.org
> >> To: freeswitch-dev at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> >>
> >> That's the same thing we saw...  Everyone wants SS7 but no one wants
> >> to help fund it... And to make matters worse just getting proper
> >> documentation for ANSI SS7 is not exactly a cheap endeavor.
> >> Unfortunately telecordia and crew still want to charge thousands of
> >> dollars for the docs when the ITU decided ages ago to make those
> >> available for free... Even then its still a substantial amount of
> >> effort to get something working (even if you have active links you can
> >
> >> use to help understand whats going on)
> >>
> >> K
> >>
> >>
> >>
> >>
> >> ________________________________
> >>
> >> From: Jan Berger <janvb at live.com>
> >> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> >> <freeswitch-dev at lists.freeswitch.org>
> >> Date: Wed, 28 Jan 2009 19:30:15 +0100
> >> To: <freeswitch-dev at lists.freeswitch.org>
> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> >>
> >> This is a little like the hen and the egg...
> >>
> >> No one will support you starting to write a SS7 ++ stack, but once you
> >
> >> get going and the first stacks turns up, so will business.
> >>
> >> Jan
> >>
> >>
> >>
> >> ________________________________
> >>
> >>
> >> From: mike at jerris.com
> >> To: freeswitch-dev at lists.freeswitch.org
> >> Date: Tue, 27 Jan 2009 15:06:11 -0500
> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> >>
> >>
> >> I still see a compelling business case for a foundation supported free
> >
> >> open source set of libs for the base protocols, commercially supported
> >
> >> by multiple contributors who build higher level applications using the
> >
> >> base protocols.  In a situation like this, commercial sponsors can
> >> pool resources for certification.  Obviously anyone using a modified
> >> version would not be using certified code but the sponsors who pay for
> >
> >> certification would still get multiple benefits.
> >>
> >> Mike
> >>
> >> On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
> >>
> >>
> >>
> >> I don't see a non Commercial one getting certification and then
> >> getting widely used here in the states... That's the problem you have
> >> to spend tons of time to get them certified... Its not like SIP where
> >> they will pretty much take anything and everything unfortunately
> >>
> >> K
> >>
> >> From: Brian West <brian at freeswitch.org>
> >> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> >> <freeswitch-dev at lists.freeswitch.org>
> >> Date: Tue, 27 Jan 2009 12:59:18 -0600
> >> To: "freeswitch-dev at lists.freeswitch.org"
> >> <freeswitch-dev at lists.freeswitch.org>
> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
> >>
> >> We should have an open source one too thats MPL or BSD ... I don't
> >> agree with selling protocols like this commercially.  Guess thats just
> >
> >> the Open Source in me wanting that stuff to be free.
> >>
> >> /b
> >>
> >> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
> >>
> >> There are things available for this in the commercial space.  One from
> >
> >> CometSig.com and one from sangoma.  Nothing at this time in the
> >> opensource arena. (OpenSS7 will not be integrated with freeswitch due
> >> to licensing
> >> issues)
> >>
> >> Contact me off list if you are interested in either of the commercial
> >> versions I can assist you with these things.
> >>
> >> Ken
> >>
> >>
> >> ________________________________
> >> check out the rest of the Windows Live(tm). More than mail-Windows
> > Live(tm)
> >> goes way beyond your inbox. More than messages
> >> <http://www.microsoft.com/windows/windowslive/>
> >> ________________________________
> >> _______________________________________________
> >> Freeswitch-dev mailing list
> >> Freeswitch-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >>
> >> ________________________________
> >> check out the rest of the Windows Live(tm). More than mail-Windows
> > Live(tm)
> >> goes way beyond your inbox. More than messages
> >> <http://www.microsoft.com/windows/windowslive/>
> >> ________________________________
> >>
> >> _______________________________________________
> >> Freeswitch-dev mailing list
> >> Freeswitch-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> http://www.freeswitch.org
> >>
> >> ________________________________
> >> Get news, entertainment and everything you care about at Live.com.
> >> Check it out! _______________________________________________
> >> Freeswitch-dev mailing list
> >> Freeswitch-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-dev mailing list
> >> Freeswitch-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
-- 
William King
Cell: 253-686-5518
E-mail: william at channelxstream.com
Channel XStream
www.channelxstream.com
1-877-600-6786

If there is a possibility that any information in our conversation might
be considered 'private' or 'sensitive' such as passwords, account 
information, legal or financial information, or anything else that you
would consider 'private' or 'sensitive' communications. It is better to 
always err on the side of security. Please encrypt the e-mail using 
my gpg key: 95C9D5B3.

If you are unfamiliar with e-mail encryption feel free to let me know 
and I can help you establish the proper protocols and procedures.
https://help.ubuntu.com/community/GnuPrivacyGuardHowto

Get my gpg key: gpg --recv-key --keyserver keyserver.ubuntu.com 95C9D5B3
Key Fingerprint: EA6F B2EE 1846 55D4 FFD9  80BA 6489 B48C 95C9 D5B3
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From msc at freeswitch.org  Wed Jan 28 15:51:57 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 28 Jan 2009 15:51:57 -0800
Subject: [Freeswitch-dev] FreeSWITCH SS7 stack
In-Reply-To: <1233185060.22743.1.camel@quentusrex-desktop>
References: <BAY125-W27F7B5DA0459D16F097397D2C80@phx.gbl>
	<C5A60FC2.4CD8A%krice@freeswitch.org>
	<BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<F20767C9-528B-4A54-A19F-8625C5AE6F7F@metrostat.net>
	<11372C8B9E603F4FACDE6AB18256DEC695A931@srvmtel.office.mtel.nl>
	<87f2f3b90901281323n28f4076eyffdfb961986eacf8@mail.gmail.com>
	<11372C8B9E603F4FACDE6AB18256DEC695A933@srvmtel.office.mtel.nl>
	<87f2f3b90901281507v43c0c8acnb59ef6fe2b9518af@mail.gmail.com>
	<1233185060.22743.1.camel@quentusrex-desktop>
Message-ID: <87f2f3b90901281551m46a80445kd415a047c605a090@mail.gmail.com>

Why thank you, sir!
-MC

On Wed, Jan 28, 2009 at 3:24 PM, William King
<william at channelxstream.com> wrote:
> Way to go MikeC, taking a lively discussion on the e-mail list serve and
> converting it into useful wiki docs. :)
>
> -William
>
> On Wed, 2009-01-28 at 15:07 -0800, Michael Collins wrote:
>> This is very helpful, thanks!
>> -MC
>>
>> On Wed, Jan 28, 2009 at 3:01 PM, Remko Kloosterman
>> <R.Kloosterman at mtel.nl> wrote:
>> > Well, I cannot disclose the commercial stuff, but I can share some
>> > experience and resources that I've found to be useful. For example:
>> >
>> > Getting started: http://pt.com/page/tutorials/ss7-tutorial/
>> > ITU recommendations: http://eu.sabotage.org/, dig in to www/ITU.
>> > I guess you know www.openss7.org
>> > Online (but recently restricted) MTP3 decoder:
>> > http://www.linkbit.com/decoder/decoder.html
>> > Perhaps some design inspiration:
>> > http://www.nmscommunications.com/DevPlatforms/Support/SWandDoc/default.h
>> > tm?section=4&ID=43&PAC=472&ST=Current&dok=
>> >
>> > Want more?
>> > Remko
>> >
>> >
>> > -----Oorspronkelijk bericht-----
>> > Van: freeswitch-dev-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Namens Michael
>> > Collins
>> > Verzonden: woensdag 28 januari 2009 22:23
>> > Aan: freeswitch-dev at lists.freeswitch.org
>> > Onderwerp: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>> >
>> > How about some SS7 documentation? ;)
>> > -MC
>> >
>> > On Wed, Jan 28, 2009 at 12:47 PM, Remko Kloosterman
>> > <R.Kloosterman at mtel.nl> wrote:
>> >> I've done some SS7 development in the past, using dedicated TX3220 and
>> >
>> >> TX4000 signalling boards from NMS Communications (recentlty aquired by
>> >
>> >> Dialogic). Even with a lot of commercial documentation available, it's
>> >
>> >> horribly complex stuff, especially redundancy. And you need another
>> >> SS7 switch to test against for interoperability. And SIP trunks are
>> >> becoming more and more commonly available these days, so one can
>> >> question if it's worth the effort. Still, I would really like it if
>> >> someone would try and develop such a feature.
>> >>
>> >> Off-topic: If you need SS7 and are willing to pay for it, I can
>> >> recommend audiocodes mediant gateways. There's a model available with
>> >> embedded SS7 signalling, provided by Teles. And real cheap too,
>> >> compared to other SS7 switches.
>> >>
>> >> ________________________________
>> >> Van: freeswitch-dev-bounces at lists.freeswitch.org
>> >> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Namens Shane
>> >> Burrell
>> >> Verzonden: woensdag 28 januari 2009 20:50
>> >> Aan: freeswitch-dev at lists.freeswitch.org
>> >> Onderwerp: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>> >>
>> >> Trust me AT&T doesn't have free docs. There are hundreds of ANSI SS7
>> >> docs and they put alot of work and money to make sure you can't get
>> > them via
>> >> google search.   Much of it is not in ITU but many ITU switch support
>> > ANSI
>> >> features such as Class 5.  Even with the ITU docs in hand you'll need
>> >> a large team to get it complete enough for Class 4/5 operations
>> > anytime soon.
>> >> On Jan 28, 2009, at 2:42 PM, Jan Berger wrote:
>> >>
>> >> Next time try to search if AT&T have a free version of the doc first -
>> >
>> >> you might get a suprice.
>> >>
>> >> ________________________________
>> >> Date: Wed, 28 Jan 2009 13:36:02 -0600
>> >> From: krice at freeswitch.org
>> >> To: freeswitch-dev at lists.freeswitch.org
>> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>> >>
>> >> Must be nice ;)
>> >>
>> >> Here in the states they want you to pay for documentation on cable
>> >> pinouts and don't even think about getting documentation for something
>> >
>> >> like TCAP LIDB without giving up your first born and an arm.
>> >>
>> >>
>> >>
>> >> ________________________________
>> >> From: Jan Berger <janvb at live.com>
>> >> Reply-To: "freeswitch-dev at lists.freeswitch.org"
>> >> <freeswitch-dev at lists.freeswitch.org>
>> >> Date: Wed, 28 Jan 2009 20:22:38 +0100
>> >> To: <freeswitch-dev at lists.freeswitch.org>
>> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>> >>
>> >> I live in Europe - and ITU/ETSI are free :)
>> >>
>> >>
>> >> ________________________________
>> >>
>> >> Date: Wed, 28 Jan 2009 12:37:48 -0600
>> >> From: krice at freeswitch.org
>> >> To: freeswitch-dev at lists.freeswitch.org
>> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>> >>
>> >> That's the same thing we saw...  Everyone wants SS7 but no one wants
>> >> to help fund it... And to make matters worse just getting proper
>> >> documentation for ANSI SS7 is not exactly a cheap endeavor.
>> >> Unfortunately telecordia and crew still want to charge thousands of
>> >> dollars for the docs when the ITU decided ages ago to make those
>> >> available for free... Even then its still a substantial amount of
>> >> effort to get something working (even if you have active links you can
>> >
>> >> use to help understand whats going on)
>> >>
>> >> K
>> >>
>> >>
>> >>
>> >>
>> >> ________________________________
>> >>
>> >> From: Jan Berger <janvb at live.com>
>> >> Reply-To: "freeswitch-dev at lists.freeswitch.org"
>> >> <freeswitch-dev at lists.freeswitch.org>
>> >> Date: Wed, 28 Jan 2009 19:30:15 +0100
>> >> To: <freeswitch-dev at lists.freeswitch.org>
>> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>> >>
>> >> This is a little like the hen and the egg...
>> >>
>> >> No one will support you starting to write a SS7 ++ stack, but once you
>> >
>> >> get going and the first stacks turns up, so will business.
>> >>
>> >> Jan
>> >>
>> >>
>> >>
>> >> ________________________________
>> >>
>> >>
>> >> From: mike at jerris.com
>> >> To: freeswitch-dev at lists.freeswitch.org
>> >> Date: Tue, 27 Jan 2009 15:06:11 -0500
>> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>> >>
>> >>
>> >> I still see a compelling business case for a foundation supported free
>> >
>> >> open source set of libs for the base protocols, commercially supported
>> >
>> >> by multiple contributors who build higher level applications using the
>> >
>> >> base protocols.  In a situation like this, commercial sponsors can
>> >> pool resources for certification.  Obviously anyone using a modified
>> >> version would not be using certified code but the sponsors who pay for
>> >
>> >> certification would still get multiple benefits.
>> >>
>> >> Mike
>> >>
>> >> On Jan 27, 2009, at 2:07 PM, Ken Rice wrote:
>> >>
>> >>
>> >>
>> >> I don't see a non Commercial one getting certification and then
>> >> getting widely used here in the states... That's the problem you have
>> >> to spend tons of time to get them certified... Its not like SIP where
>> >> they will pretty much take anything and everything unfortunately
>> >>
>> >> K
>> >>
>> >> From: Brian West <brian at freeswitch.org>
>> >> Reply-To: "freeswitch-dev at lists.freeswitch.org"
>> >> <freeswitch-dev at lists.freeswitch.org>
>> >> Date: Tue, 27 Jan 2009 12:59:18 -0600
>> >> To: "freeswitch-dev at lists.freeswitch.org"
>> >> <freeswitch-dev at lists.freeswitch.org>
>> >> Subject: Re: [Freeswitch-dev] FreeSWITCH SS7 stack
>> >>
>> >> We should have an open source one too thats MPL or BSD ... I don't
>> >> agree with selling protocols like this commercially.  Guess thats just
>> >
>> >> the Open Source in me wanting that stuff to be free.
>> >>
>> >> /b
>> >>
>> >> On Jan 27, 2009, at 12:27 PM, Ken Rice wrote:
>> >>
>> >> There are things available for this in the commercial space.  One from
>> >
>> >> CometSig.com and one from sangoma.  Nothing at this time in the
>> >> opensource arena. (OpenSS7 will not be integrated with freeswitch due
>> >> to licensing
>> >> issues)
>> >>
>> >> Contact me off list if you are interested in either of the commercial
>> >> versions I can assist you with these things.
>> >>
>> >> Ken
>> >>
>> >>
>> >> ________________________________
>> >> check out the rest of the Windows Live(tm). More than mail-Windows
>> > Live(tm)
>> >> goes way beyond your inbox. More than messages
>> >> <http://www.microsoft.com/windows/windowslive/>
>> >> ________________________________
>> >> _______________________________________________
>> >> Freeswitch-dev mailing list
>> >> Freeswitch-dev at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> http://www.freeswitch.org <http://www.freeswitch.org/>
>> >>
>> >> ________________________________
>> >> check out the rest of the Windows Live(tm). More than mail-Windows
>> > Live(tm)
>> >> goes way beyond your inbox. More than messages
>> >> <http://www.microsoft.com/windows/windowslive/>
>> >> ________________________________
>> >>
>> >> _______________________________________________
>> >> Freeswitch-dev mailing list
>> >> Freeswitch-dev at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> http://www.freeswitch.org
>> >>
>> >> ________________________________
>> >> Get news, entertainment and everything you care about at Live.com.
>> >> Check it out! _______________________________________________
>> >> Freeswitch-dev mailing list
>> >> Freeswitch-dev at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> Freeswitch-dev mailing list
>> >> Freeswitch-dev at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> > _______________________________________________
>> > Freeswitch-dev mailing list
>> > Freeswitch-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > Freeswitch-dev mailing list
>> > Freeswitch-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
> --
> William King
> Cell: 253-686-5518
> E-mail: william at channelxstream.com
> Channel XStream
> www.channelxstream.com
> 1-877-600-6786
>
> If there is a possibility that any information in our conversation might
> be considered 'private' or 'sensitive' such as passwords, account
> information, legal or financial information, or anything else that you
> would consider 'private' or 'sensitive' communications. It is better to
> always err on the side of security. Please encrypt the e-mail using
> my gpg key: 95C9D5B3.
>
> If you are unfamiliar with e-mail encryption feel free to let me know
> and I can help you establish the proper protocols and procedures.
> https://help.ubuntu.com/community/GnuPrivacyGuardHowto
>
> Get my gpg key: gpg --recv-key --keyserver keyserver.ubuntu.com 95C9D5B3
> Key Fingerprint: EA6F B2EE 1846 55D4 FFD9  80BA 6489 B48C 95C9 D5B3
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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From regs at kinetix.gr  Thu Jan 29 04:40:11 2009
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Thu, 29 Jan 2009 14:40:11 +0200
Subject: [Freeswitch-dev] freeswitch development question (memory
	allocation?)
Message-ID: <4981A3AB.2000803@kinetix.gr>

I have the code below :

struct radacct_thread_handle {
        switch_core_session_t *session;
        switch_mutex_t *mutex;
        switch_thread_cond_t *cond;
};

static switch_status_t my_on_routing(switch_core_session_t *session){

        switch_thread_t *thread;
        switch_threadattr_t *thd_attr = NULL;     

        switch_memory_pool_t *pool;

        struct radacct_thread_handle *thread_params = NULL;
       
        pool = switch_core_session_get_pool(session);     

        thread_params->session = session;
       
        ...
       
}

when the program reaches the last line (thread_params->session = session;)
I get a core dump. Is this a memory allocation error? Is it because I am 
making
use of the wrong pool? Please enlighten me because I am not an experienced c
programmer, and I am struggling to get familiar with the FS API.

-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 



From regs at kinetix.gr  Thu Jan 29 05:00:38 2009
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Thu, 29 Jan 2009 15:00:38 +0200
Subject: [Freeswitch-dev] freeswitch development question (memory
 allocation?) [SOLVED]
In-Reply-To: <4981A3AB.2000803@kinetix.gr>
References: <4981A3AB.2000803@kinetix.gr>
Message-ID: <4981A876.5030605@kinetix.gr>

I found it out by myself!
 (why is it that we always come with the solution right after posting to 
the list?)

I inserted :

thread_params = switch_core_session_alloc(session, sizeof(*thread_params));

before the pool initialization.

But still,  can I get some answers to the questions bellow about
how to effectively handle memory allocations?



Apostolos Pantsiopoulos wrote:
> I have the code below :
>
> struct radacct_thread_handle {
>         switch_core_session_t *session;
>         switch_mutex_t *mutex;
>         switch_thread_cond_t *cond;
> };
>
> static switch_status_t my_on_routing(switch_core_session_t *session){
>
>         switch_thread_t *thread;
>         switch_threadattr_t *thd_attr = NULL;     
>
>         switch_memory_pool_t *pool;
>
>         struct radacct_thread_handle *thread_params = NULL;
>        
>         pool = switch_core_session_get_pool(session);     
>
>         thread_params->session = session;
>        
>         ...
>        
> }
>
> when the program reaches the last line (thread_params->session = session;)
> I get a core dump. Is this a memory allocation error? Is it because I am 
> making
> use of the wrong pool? Please enlighten me because I am not an experienced c
> programmer, and I am struggling to get familiar with the FS API.
>
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 



From mike at jerris.com  Thu Jan 29 05:10:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 29 Jan 2009 08:10:26 -0500
Subject: [Freeswitch-dev] freeswitch development question (memory
	allocation?) [SOLVED]
In-Reply-To: <4981A876.5030605@kinetix.gr>
References: <4981A3AB.2000803@kinetix.gr> <4981A876.5030605@kinetix.gr>
Message-ID: <2EDFD0BF-7767-42C2-8B09-48A75B0B2F08@jerris.com>

I am not sure even after re-reading what your question is, could you  
try to rephrase?

Mike

On Jan 29, 2009, at 8:00 AM, Apostolos Pantsiopoulos wrote:

> I found it out by myself!
> (why is it that we always come with the solution right after posting  
> to
> the list?)
>
> I inserted :
>
> thread_params = switch_core_session_alloc(session,  
> sizeof(*thread_params));
>
> before the pool initialization.
>
> But still,  can I get some answers to the questions bellow about
> how to effectively handle memory allocations?
>
>
>
> Apostolos Pantsiopoulos wrote:
>> I have the code below :
>>
>> struct radacct_thread_handle {
>>        switch_core_session_t *session;
>>        switch_mutex_t *mutex;
>>        switch_thread_cond_t *cond;
>> };
>>
>> static switch_status_t my_on_routing(switch_core_session_t *session){
>>
>>        switch_thread_t *thread;
>>        switch_threadattr_t *thd_attr = NULL;
>>
>>        switch_memory_pool_t *pool;
>>
>>        struct radacct_thread_handle *thread_params = NULL;
>>
>>        pool = switch_core_session_get_pool(session);
>>
>>        thread_params->session = session;
>>
>>        ...
>>
>> }
>>
>> when the program reaches the last line (thread_params->session =  
>> session;)
>> I get a core dump. Is this a memory allocation error? Is it because  
>> I am
>> making
>> use of the wrong pool? Please enlighten me because I am not an  
>> experienced c
>> programmer, and I am struggling to get familiar with the FS API.
>>
>>



From regs at kinetix.gr  Thu Jan 29 05:23:27 2009
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Thu, 29 Jan 2009 15:23:27 +0200
Subject: [Freeswitch-dev] freeswitch development question
 (memory	allocation?) [SOLVED]
In-Reply-To: <2EDFD0BF-7767-42C2-8B09-48A75B0B2F08@jerris.com>
References: <4981A3AB.2000803@kinetix.gr> <4981A876.5030605@kinetix.gr>
	<2EDFD0BF-7767-42C2-8B09-48A75B0B2F08@jerris.com>
Message-ID: <4981ADCF.6050802@kinetix.gr>

The question is : which pool should I use for each different thing I am 
trying to accomplish?

For instance the code below is part of my mod_radius_cdr experimentation.
When I am making use of a pool within a module should I :

a) create a new pool?
b) make use of an existing one?
c) what kind of pool should I use?
d) is it really just one pool (accessed through different handlers) and 
I am making myself look like a fool so far?





Michael Jerris wrote:
> I am not sure even after re-reading what your question is, could you  
> try to rephrase?
>
> Mike
>
> On Jan 29, 2009, at 8:00 AM, Apostolos Pantsiopoulos wrote:
>
>   
>> I found it out by myself!
>> (why is it that we always come with the solution right after posting  
>> to
>> the list?)
>>
>> I inserted :
>>
>> thread_params = switch_core_session_alloc(session,  
>> sizeof(*thread_params));
>>
>> before the pool initialization.
>>
>> But still,  can I get some answers to the questions bellow about
>> how to effectively handle memory allocations?
>>
>>
>>
>> Apostolos Pantsiopoulos wrote:
>>     
>>> I have the code below :
>>>
>>> struct radacct_thread_handle {
>>>        switch_core_session_t *session;
>>>        switch_mutex_t *mutex;
>>>        switch_thread_cond_t *cond;
>>> };
>>>
>>> static switch_status_t my_on_routing(switch_core_session_t *session){
>>>
>>>        switch_thread_t *thread;
>>>        switch_threadattr_t *thd_attr = NULL;
>>>
>>>        switch_memory_pool_t *pool;
>>>
>>>        struct radacct_thread_handle *thread_params = NULL;
>>>
>>>        pool = switch_core_session_get_pool(session);
>>>
>>>        thread_params->session = session;
>>>
>>>        ...
>>>
>>> }
>>>
>>> when the program reaches the last line (thread_params->session =  
>>> session;)
>>> I get a core dump. Is this a memory allocation error? Is it because  
>>> I am
>>> making
>>> use of the wrong pool? Please enlighten me because I am not an  
>>> experienced c
>>> programmer, and I am struggling to get familiar with the FS API.
>>>
>>>
>>>       
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From mike at jerris.com  Thu Jan 29 05:50:31 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 29 Jan 2009 08:50:31 -0500
Subject: [Freeswitch-dev] freeswitch development question (memory
	allocation?) [SOLVED]
In-Reply-To: <4981ADCF.6050802@kinetix.gr>
References: <4981A3AB.2000803@kinetix.gr> <4981A876.5030605@kinetix.gr>
	<2EDFD0BF-7767-42C2-8B09-48A75B0B2F08@jerris.com>
	<4981ADCF.6050802@kinetix.gr>
Message-ID: <A822F192-729C-4B0C-9805-FE7E9B8BBA52@jerris.com>


On Jan 29, 2009, at 8:23 AM, Apostolos Pantsiopoulos wrote:

> The question is : which pool should I use for each different thing I  
> am trying to accomplish?
>
> For instance the code below is part of my mod_radius_cdr  
> experimentation.
> When I am making use of a pool within a module should I :
>
> a) create a new pool?

Depends what scope you need it for.
>
> b) make use of an existing one?
Depends what scope you need it for.

> c) what kind of pool should I use?

There are not different kinds of pools
>
> d) is it really just one pool (accessed through different handlers)  
> and I am making myself look like a fool so far?

Depends what you mean by it, but there are many pools.


A memory pool is an object that you can use to make allocations that  
can all be freed at a later time as a single call to pool destroy.  It  
essentially balls up all your malloc calls and tracks them on a single  
handle to be freed later.  It also has some performance benefits over  
malloc in that it allocates in larger chunks usually then dishes out  
from those chunks on demand.  Because a pool is freed all together, we  
think of pools as having a scope.  For example, each FreeSWITCH  
session has a pool, and we use that pool to allocate things that need  
to be around for the life of the session such as the strings used in  
the caller profile.  It would be horribly complex to have to go and  
clean those things up later if they were malloc'ed strings, so the  
pool makes that easy for us.  So most of your questions above depend  
on what the scope of what you are trying to allocate should be.  If it  
is something that should live for the life of a session, you should  
probably use the session pool, if it is something your just using  
inside one function, in most cases you should just stack allocate, if  
its something that needs to live for a finite time but goes through  
many functions in complex interactions, (such as a file handle and all  
its file operations) you might want to stick your own new pool in your  
structure that you can destroy when you destroy that handle.

Mike

>
>
>
>
>
>
> Michael Jerris wrote:
>>
>> I am not sure even after re-reading what your question is, could you
>> try to rephrase?
>>
>> Mike
>>
>> On Jan 29, 2009, at 8:00 AM, Apostolos Pantsiopoulos wrote:
>>
>>
>>> I found it out by myself!
>>> (why is it that we always come with the solution right after posting
>>> to
>>> the list?)
>>>
>>> I inserted :
>>>
>>> thread_params = switch_core_session_alloc(session,
>>> sizeof(*thread_params));
>>>
>>> before the pool initialization.
>>>
>>> But still,  can I get some answers to the questions bellow about
>>> how to effectively handle memory allocations?
>>>
>>>
>>>
>>> Apostolos Pantsiopoulos wrote:
>>>
>>>> I have the code below :
>>>>
>>>> struct radacct_thread_handle {
>>>>        switch_core_session_t *session;
>>>>        switch_mutex_t *mutex;
>>>>        switch_thread_cond_t *cond;
>>>> };
>>>>
>>>> static switch_status_t my_on_routing(switch_core_session_t  
>>>> *session){
>>>>
>>>>        switch_thread_t *thread;
>>>>        switch_threadattr_t *thd_attr = NULL;
>>>>
>>>>        switch_memory_pool_t *pool;
>>>>
>>>>        struct radacct_thread_handle *thread_params = NULL;
>>>>
>>>>        pool = switch_core_session_get_pool(session);
>>>>
>>>>        thread_params->session = session;
>>>>
>>>>        ...
>>>>
>>>> }
>>>>
>>>> when the program reaches the last line (thread_params->session =
>>>> session;)
>>>> I get a core dump. Is this a memory allocation error? Is it because
>>>> I am
>>>> making
>>>> use of the wrong pool? Please enlighten me because I am not an
>>>> experienced c
>>>> programmer, and I am struggling to get familiar with the FS API.
>>>>
>>>>
>>>>
>>


From anthony.minessale at gmail.com  Thu Jan 29 05:55:36 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 29 Jan 2009 07:55:36 -0600
Subject: [Freeswitch-dev] freeswitch development question (memory
	allocation?) [SOLVED]
In-Reply-To: <4981ADCF.6050802@kinetix.gr>
References: <4981A3AB.2000803@kinetix.gr> <4981A876.5030605@kinetix.gr>
	<2EDFD0BF-7767-42C2-8B09-48A75B0B2F08@jerris.com>
	<4981ADCF.6050802@kinetix.gr>
Message-ID: <191c3a030901290555i58186cc8p2cdecd7b700f33dc@mail.gmail.com>

if your intent is to keep the data around longer than the session in it's
own thread, you will want to make a new pool
and use that pool to launch the thread.  pass the pool into the thread as a
member of your thread private data and free
the pool before you exit the thread.

You cannot bring the session with you into the thread unless you read lock
the session first so it will not be destroyed while
you are using it in the thread.

Be aware keeping the session around in the thread just a wild goose chase
from doing it the why it already was doing it.
The session thread is already independent and it does not stop any other
call from proceeding so you may as well do all the work
in your session thread in the state change handler like it already does and
just add more strict timeout.


On Thu, Jan 29, 2009 at 7:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>  The question is : which pool should I use for each different thing I am
> trying to accomplish?
>
> For instance the code below is part of my mod_radius_cdr experimentation.
> When I am making use of a pool within a module should I :
>
> a) create a new pool?
> b) make use of an existing one?
> c) what kind of pool should I use?
> d) is it really just one pool (accessed through different handlers) and I
> am making myself look like a fool so far?
>
>
>
>
>
>
> Michael Jerris wrote:
>
> I am not sure even after re-reading what your question is, could you
> try to rephrase?
>
> Mike
>
> On Jan 29, 2009, at 8:00 AM, Apostolos Pantsiopoulos wrote:
>
>
>
>  I found it out by myself!
> (why is it that we always come with the solution right after posting
> to
> the list?)
>
> I inserted :
>
> thread_params = switch_core_session_alloc(session,
> sizeof(*thread_params));
>
> before the pool initialization.
>
> But still,  can I get some answers to the questions bellow about
> how to effectively handle memory allocations?
>
>
>
> Apostolos Pantsiopoulos wrote:
>
>
>  I have the code below :
>
> struct radacct_thread_handle {
>        switch_core_session_t *session;
>        switch_mutex_t *mutex;
>        switch_thread_cond_t *cond;
> };
>
> static switch_status_t my_on_routing(switch_core_session_t *session){
>
>        switch_thread_t *thread;
>        switch_threadattr_t *thd_attr = NULL;
>
>        switch_memory_pool_t *pool;
>
>        struct radacct_thread_handle *thread_params = NULL;
>
>        pool = switch_core_session_get_pool(session);
>
>        thread_params->session = session;
>
>        ...
>
> }
>
> when the program reaches the last line (thread_params->session =
> session;)
> I get a core dump. Is this a memory allocation error? Is it because
> I am
> making
> use of the wrong pool? Please enlighten me because I am not an
> experienced c
> programmer, and I am struggling to get familiar with the FS API.
>
>
>
>
>  _______________________________________________
> Freeswitch-dev mailing listFreeswitch-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From regs at kinetix.gr  Thu Jan 29 07:13:36 2009
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Thu, 29 Jan 2009 17:13:36 +0200
Subject: [Freeswitch-dev] freeswitch development question
 (memory	allocation?) [SOLVED]
In-Reply-To: <191c3a030901290555i58186cc8p2cdecd7b700f33dc@mail.gmail.com>
References: <4981A3AB.2000803@kinetix.gr>
	<4981A876.5030605@kinetix.gr>	<2EDFD0BF-7767-42C2-8B09-48A75B0B2F08@jerris.com>	<4981ADCF.6050802@kinetix.gr>
	<191c3a030901290555i58186cc8p2cdecd7b700f33dc@mail.gmail.com>
Message-ID: <4981C7A0.7090000@kinetix.gr>

Thanks for the explanations (both Anthony and Michael).
You both clarified the subject quite well.

Anthony Minessale wrote:
> if your intent is to keep the data around longer than the session in 
> it's own thread, you will want to make a new pool
> and use that pool to launch the thread.  pass the pool into the thread 
> as a member of your thread private data and free
> the pool before you exit the thread.
>
> You cannot bring the session with you into the thread unless you read 
> lock the session first so it will not be destroyed while
> you are using it in the thread.
>
> Be aware keeping the session around in the thread just a wild goose 
> chase from doing it the why it already was doing it.
> The session thread is already independent and it does not stop any 
> other call from proceeding so you may as well do all the work
> in your session thread in the state change handler like it already 
> does and just add more strict timeout.
>  
>
> On Thu, Jan 29, 2009 at 7:23 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     The question is : which pool should I use for each different thing
>     I am trying to accomplish?
>
>     For instance the code below is part of my mod_radius_cdr
>     experimentation.
>     When I am making use of a pool within a module should I :
>
>     a) create a new pool?
>     b) make use of an existing one?
>     c) what kind of pool should I use?
>     d) is it really just one pool (accessed through different
>     handlers) and I am making myself look like a fool so far?
>
>
>
>
>
>
>     Michael Jerris wrote:
>>     I am not sure even after re-reading what your question is, could you  
>>     try to rephrase?
>>
>>     Mike
>>
>>     On Jan 29, 2009, at 8:00 AM, Apostolos Pantsiopoulos wrote:
>>
>>       
>>>     I found it out by myself!
>>>     (why is it that we always come with the solution right after posting  
>>>     to
>>>     the list?)
>>>
>>>     I inserted :
>>>
>>>     thread_params = switch_core_session_alloc(session,  
>>>     sizeof(*thread_params));
>>>
>>>     before the pool initialization.
>>>
>>>     But still,  can I get some answers to the questions bellow about
>>>     how to effectively handle memory allocations?
>>>
>>>
>>>
>>>     Apostolos Pantsiopoulos wrote:
>>>         
>>>>     I have the code below :
>>>>
>>>>     struct radacct_thread_handle {
>>>>            switch_core_session_t *session;
>>>>            switch_mutex_t *mutex;
>>>>            switch_thread_cond_t *cond;
>>>>     };
>>>>
>>>>     static switch_status_t my_on_routing(switch_core_session_t *session){
>>>>
>>>>            switch_thread_t *thread;
>>>>            switch_threadattr_t *thd_attr = NULL;
>>>>
>>>>            switch_memory_pool_t *pool;
>>>>
>>>>            struct radacct_thread_handle *thread_params = NULL;
>>>>
>>>>            pool = switch_core_session_get_pool(session);
>>>>
>>>>            thread_params->session = session;
>>>>
>>>>            ...
>>>>
>>>>     }
>>>>
>>>>     when the program reaches the last line (thread_params->session =  
>>>>     session;)
>>>>     I get a core dump. Is this a memory allocation error? Is it because  
>>>>     I am
>>>>     making
>>>>     use of the wrong pool? Please enlighten me because I am not an  
>>>>     experienced c
>>>>     programmer, and I am struggling to get familiar with the FS API.
>>>>
>>>>
>>>>           
>>     _______________________________________________
>>     Freeswitch-dev mailing list
>>     Freeswitch-dev at lists.freeswitch.org <mailto:Freeswitch-dev at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>     http://www.freeswitch.org
>>       
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
>     _______________________________________________
>     Freeswitch-dev mailing list
>     Freeswitch-dev at lists.freeswitch.org
>     <mailto:Freeswitch-dev at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From mike at jerris.com  Thu Jan 29 07:22:16 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 29 Jan 2009 10:22:16 -0500
Subject: [Freeswitch-dev] freeswitch development question (memory
	allocation?) [SOLVED]
In-Reply-To: <4981C7A0.7090000@kinetix.gr>
References: <4981A3AB.2000803@kinetix.gr>
	<4981A876.5030605@kinetix.gr>	<2EDFD0BF-7767-42C2-8B09-48A75B0B2F08@jerris.com>	<4981ADCF.6050802@kinetix.gr>
	<191c3a030901290555i58186cc8p2cdecd7b700f33dc@mail.gmail.com>
	<4981C7A0.7090000@kinetix.gr>
Message-ID: <D1F5E873-8C0D-4CD5-83D1-1D43B4EE8E5F@jerris.com>

would you mind getting some of this information into developer pages  
on the wiki?

http://wiki.freeswitch.org/wiki/Documentation/Developer_Documentation
http://wiki.freeswitch.org/wiki/Developer_Potpourri


Thanks
Mike

On Jan 29, 2009, at 10:13 AM, Apostolos Pantsiopoulos wrote:

> Thanks for the explanations (both Anthony and Michael).
> You both clarified the subject quite well.
>
> Anthony Minessale wrote:
>>
>> if your intent is to keep the data around longer than the session  
>> in it's own thread, you will want to make a new pool
>> and use that pool to launch the thread.  pass the pool into the  
>> thread as a member of your thread private data and free
>> the pool before you exit the thread.
>>
>> You cannot bring the session with you into the thread unless you  
>> read lock the session first so it will not be destroyed while
>> you are using it in the thread.
>>
>> Be aware keeping the session around in the thread just a wild goose  
>> chase from doing it the why it already was doing it.
>> The session thread is already independent and it does not stop any  
>> other call from proceeding so you may as well do all the work
>> in your session thread in the state change handler like it already  
>> does and just add more strict timeout.
>>
>>
>> On Thu, Jan 29, 2009 at 7:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr 
>> > wrote:
>> The question is : which pool should I use for each different thing  
>> I am trying to accomplish?
>>
>> For instance the code below is part of my mod_radius_cdr  
>> experimentation.
>> When I am making use of a pool within a module should I :
>>
>> a) create a new pool?
>> b) make use of an existing one?
>> c) what kind of pool should I use?
>> d) is it really just one pool (accessed through different handlers)  
>> and I am making myself look like a fool so far?
>>
>>
>>
>>
>>
>>
>> Michael Jerris wrote:
>>>
>>> I am not sure even after re-reading what your question is, could you
>>> try to rephrase?
>>>
>>> Mike
>>>
>>> On Jan 29, 2009, at 8:00 AM, Apostolos Pantsiopoulos wrote:
>>>
>>>
>>>> I found it out by myself!
>>>> (why is it that we always come with the solution right after  
>>>> posting
>>>> to
>>>> the list?)
>>>>
>>>> I inserted :
>>>>
>>>> thread_params = switch_core_session_alloc(session,
>>>> sizeof(*thread_params));
>>>>
>>>> before the pool initialization.
>>>>
>>>> But still,  can I get some answers to the questions bellow about
>>>> how to effectively handle memory allocations?
>>>>
>>>>
>>>>
>>>> Apostolos Pantsiopoulos wrote:
>>>>
>>>>> I have the code below :
>>>>>
>>>>> struct radacct_thread_handle {
>>>>>        switch_core_session_t *session;
>>>>>        switch_mutex_t *mutex;
>>>>>        switch_thread_cond_t *cond;
>>>>> };
>>>>>
>>>>> static switch_status_t my_on_routing(switch_core_session_t  
>>>>> *session){
>>>>>
>>>>>        switch_thread_t *thread;
>>>>>        switch_threadattr_t *thd_attr = NULL;
>>>>>
>>>>>        switch_memory_pool_t *pool;
>>>>>
>>>>>        struct radacct_thread_handle *thread_params = NULL;
>>>>>
>>>>>        pool = switch_core_session_get_pool(session);
>>>>>
>>>>>        thread_params->session = session;
>>>>>
>>>>>        ...
>>>>>
>>>>> }
>>>>>
>>>>> when the program reaches the last line (thread_params->session =
>>>>> session;)
>>>>> I get a core dump. Is this a memory allocation error? Is it  
>>>>> because
>>>>> I am
>>>>> making
>>>>> use of the wrong pool? Please enlighten me because I am not an
>>>>> experienced c
>>>>> programmer, and I am struggling to get familiar with the FS API.
>>>>>

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From regs at kinetix.gr  Fri Jan 30 00:02:14 2009
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 30 Jan 2009 10:02:14 +0200
Subject: [Freeswitch-dev] freeswitch development question
 (memory	allocation?) [SOLVED]
In-Reply-To: <4981C7A0.7090000@kinetix.gr>
References: <4981A3AB.2000803@kinetix.gr>	<4981A876.5030605@kinetix.gr>	<2EDFD0BF-7767-42C2-8B09-48A75B0B2F08@jerris.com>	<4981ADCF.6050802@kinetix.gr>	<191c3a030901290555i58186cc8p2cdecd7b700f33dc@mail.gmail.com>
	<4981C7A0.7090000@kinetix.gr>
Message-ID: <4982B406.1050107@kinetix.gr>

Hi,

One more clarification please :

In order for my thread to use the session data before the session is 
destroyed I am using
my own mutex and condition signal. e.g.

(pseudo code follows) :

thread_function(switch_thread_t *thread, void *ptr){

// the pointer ptr is a struct that contains a pointer to session,the 
mutex and condition created in the calling thread

(session dependend stuff)

switch_thread_cond_signal(sth->cond); //sends signal to calling thread 
that we stopped using the session so it can exit

(session independent stuff)

}
calling_function(switch_core_session_t *session){

       (init stuff)

        switch_mutex_init(&thread_params->mutex, SWITCH_MUTEX_NESTED, pool);
        switch_thread_cond_create(&thread_params->cond, pool);

       (thread creation stuff)

        switch_mutex_lock( thread_params->mutex );

        switch_thread_cond_wait(thread_params->cond, 
thread_params->mutex); // this is were the calling thread waits for a signal

        switch_mutex_unlock( thread_params->mutex );

        return SWITCH_STATUS_SUCCESS;

}

Is the above correct? Is there another way to create a "wait" state for 
the calling thread until the child thread
is done with the session data?




Apostolos Pantsiopoulos wrote:
> Thanks for the explanations (both Anthony and Michael).
> You both clarified the subject quite well.
>
> Anthony Minessale wrote:
>> if your intent is to keep the data around longer than the session in 
>> it's own thread, you will want to make a new pool
>> and use that pool to launch the thread.  pass the pool into the 
>> thread as a member of your thread private data and free
>> the pool before you exit the thread.
>>
>> You cannot bring the session with you into the thread unless you read 
>> lock the session first so it will not be destroyed while
>> you are using it in the thread.
>>
>> Be aware keeping the session around in the thread just a wild goose 
>> chase from doing it the why it already was doing it.
>> The session thread is already independent and it does not stop any 
>> other call from proceeding so you may as well do all the work
>> in your session thread in the state change handler like it already 
>> does and just add more strict timeout.
>>  
>>
>> On Thu, Jan 29, 2009 at 7:23 AM, Apostolos Pantsiopoulos 
>> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>>
>>     The question is : which pool should I use for each different
>>     thing I am trying to accomplish?
>>
>>     For instance the code below is part of my mod_radius_cdr
>>     experimentation.
>>     When I am making use of a pool within a module should I :
>>
>>     a) create a new pool?
>>     b) make use of an existing one?
>>     c) what kind of pool should I use?
>>     d) is it really just one pool (accessed through different
>>     handlers) and I am making myself look like a fool so far?
>>
>>
>>
>>
>>
>>
>>     Michael Jerris wrote:
>>>     I am not sure even after re-reading what your question is, could you  
>>>     try to rephrase?
>>>
>>>     Mike
>>>
>>>     On Jan 29, 2009, at 8:00 AM, Apostolos Pantsiopoulos wrote:
>>>
>>>       
>>>>     I found it out by myself!
>>>>     (why is it that we always come with the solution right after posting  
>>>>     to
>>>>     the list?)
>>>>
>>>>     I inserted :
>>>>
>>>>     thread_params = switch_core_session_alloc(session,  
>>>>     sizeof(*thread_params));
>>>>
>>>>     before the pool initialization.
>>>>
>>>>     But still,  can I get some answers to the questions bellow about
>>>>     how to effectively handle memory allocations?
>>>>
>>>>
>>>>
>>>>     Apostolos Pantsiopoulos wrote:
>>>>         
>>>>>     I have the code below :
>>>>>
>>>>>     struct radacct_thread_handle {
>>>>>            switch_core_session_t *session;
>>>>>            switch_mutex_t *mutex;
>>>>>            switch_thread_cond_t *cond;
>>>>>     };
>>>>>
>>>>>     static switch_status_t my_on_routing(switch_core_session_t *session){
>>>>>
>>>>>            switch_thread_t *thread;
>>>>>            switch_threadattr_t *thd_attr = NULL;
>>>>>
>>>>>            switch_memory_pool_t *pool;
>>>>>
>>>>>            struct radacct_thread_handle *thread_params = NULL;
>>>>>
>>>>>            pool = switch_core_session_get_pool(session);
>>>>>
>>>>>            thread_params->session = session;
>>>>>
>>>>>            ...
>>>>>
>>>>>     }
>>>>>
>>>>>     when the program reaches the last line (thread_params->session =  
>>>>>     session;)
>>>>>     I get a core dump. Is this a memory allocation error? Is it because  
>>>>>     I am
>>>>>     making
>>>>>     use of the wrong pool? Please enlighten me because I am not an  
>>>>>     experienced c
>>>>>     programmer, and I am struggling to get familiar with the FS API.
>>>>>
>>>>>
>>>>>           
>>>     _______________________________________________
>>>     Freeswitch-dev mailing list
>>>     Freeswitch-dev at lists.freeswitch.org <mailto:Freeswitch-dev at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>     http://www.freeswitch.org
>>>       
>>
>>
>>     -- 
>>     -------------------------------------------
>>     Apostolos Pantsiopoulos
>>     Kinetix Tele.com R & D
>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>     ------------------------------------------- 
>>
>>
>>     _______________________________________________
>>     Freeswitch-dev mailing list
>>     Freeswitch-dev at lists.freeswitch.org
>>     <mailto:Freeswitch-dev at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>     http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com 
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org 
>> <mailto:sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888 
>> <http://iax:guest at conference.freeswitch.org/888>
>> googletalk:conf+888 at conference.freeswitch.org 
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>   
>
>
> -- 
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> ------------------------------------------- 
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From hselasky at c2i.net  Fri Jan 30 02:33:29 2009
From: hselasky at c2i.net (Hans Petter Selasky)
Date: Fri, 30 Jan 2009 11:33:29 +0100
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters - status
In-Reply-To: <BAY119-W3ABD90DFEA06701EE0292D2C80@phx.gbl>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<C5A611E3.4CD8D%krice@freeswitch.org>
	<BAY119-W3ABD90DFEA06701EE0292D2C80@phx.gbl>
Message-ID: <200901301133.30826.hselasky@c2i.net>

Hi,

I'm currently working on ISDN (E1/T1/BRI) support, D-channel protocol wise and 
hardware wise in connection to FreeSwitch. I'm also adding support for analog 
phone adapters through the same API. I promised to Mike Jerris several times 
I would do something, but then the time ran away ... This week I put down 
several hours integrating the ISDN support in FreeSwitch with the ISDN 
support in ISDN4BSD, which should end up with a unified BSD-licensed OpenZap 
code base.

If you are interested in giving some feedback you can check out the following 
files:

svn --username anonsvn --password anonsvn \
      checkout svn://svn.turbocat.net/i4b/trunk/openzap.hps

The files that are there are mainly for inter-process communication, and have 
been constructed in a way that allows easy porting to other platforms. In 
general sockets can be used for this on other UNIX compatible systems as a 
fallback mechanism where the optimised kernel mechanism is not present.

The core (FreeBSD kernel module): Handles routing and broadcasting.
The client (FreeBSD library): Dumb node sending and receiving messages.

What I plan next is to implement a set of nodes, like [ISDN] controller-  
Q921- and Q931- node. Nodes have pre-defined address ranges which they are 
listening to. Hence I already have a DSS1 stack, splitting it up into using 
message based communication should not be a very big task.

Then you have the application, like FreeSwitch being a node aswell picking up 
broadcast events, which are mostly incoming calls from Q931 and making 
outgoing calls.

Messages have been split into two categories: Important-frames and 
Unimportant-frames. I-Frames gets to use 64 queue entries before getting 
dropped. U-Frames gets to use only 32 queue entries before getting dropped. 
These numbers have not been tuned yet.

Also I have thought through message timing. In bigger systems I see that it is 
required to dither messages in time. On cannot simply receive 30x400 bytes on 
an E1 and burst it into the system, because the message queues will overflow 
pretty quickly. Instead the interrupt rate of the hardware needs to be 
increased so that 400 bytes are received in "1/30 * interval" fashion. This 
will generally improve the system and load-balance in time.


The generic Message Header looks like this:

/* NOTE: All fields are little endian */

struct zap_hdr {
        uint8_t dwDstNode[4];           /* Destination Node */
        uint8_t dwSrcNode[4];           /* Source Node */
        uint8_t wCommand[2];            /* Command Number */
        uint8_t wMsgNum[2];             /* Message Number */
        uint8_t bReserved[4];           /* Reserved bytes */
};

If you have access to ISDN testing equipment and would like to help test the 
non-EuroISDN variants of my upcoming ISDN implementation, please let me know.

I expect to have something up and running within the next month, hopefully not 
running out of time this time :-)

--HPS


From regs at kinetix.gr  Fri Jan 30 03:14:07 2009
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 30 Jan 2009 13:14:07 +0200
Subject: [Freeswitch-dev] mod_radius_cdr behavior and questions
Message-ID: <4982E0FF.8070805@kinetix.gr>

Hi,

    I am tweaking the mod_radius_cdr module to archive the behavior
that most NASes have (i.e. accounting packets are sent in a separate
thread so that the submission does not interfere with the execution of
the call). While doing that, however, I bumped into another behavior of
the module that I think is not desirable (at least by me) :

    While on a bridge, the module sends one acct start packet at the 
beginning of the originating
leg (on_routing event handler) and two acct stop packets at the end of 
each leg
(inbound and outbound). My opinion is that it should send one accounting 
start
packet at the beginning of each call leg (inbound, outbound) resulting 
to a total
number of two acct start packets. It is generally accepted that acct 
start/stop packets
come in pairs so that billing applications can handle them accordingly.
   
    Some NAS's radius radius implementations have some other 
configuration modes
like the Cisco's RADIUS Packet Suppression. When in this mode the Cisco NAS
sends only an accounting start/stop pair at the end of a final dialpeer 
attempt (and suppresses
all the previous failed dialpeer attempts) thus resulting to less 
network traffic. Other
NASes (such as MERA MVTS) can send a start/stop pair for each leg OR a 
start/stop
pair for each end to end call, depending on the configuration. Opensips 
follows
the star/stop pair by call leg paradigm. No matter what the 
implementation, all of them
always send a acct start/stop pair. This is a common thing. And all the 
billing platforms
can deal only with paired start/stops.

    The current module behavior (one start two stops) can complicate 
things since the
radius server would not know how to match the second stop packet with 
its equivalent start.

    Before I get the infamous answer "pathes welcomed" I would like to 
state that I am
already involved in changing this behavior (through a patch). So my real 
question is this :

I noticed that the module uses the switch_core_add_state_handler 
function to register
its handler table :

switch_core_add_state_handler(&state_handlers);

So the on_routing ( or the on_execute) event happens only when the 
inbound call is started.
When the outbound call is initiated no handler is available to hook up a 
function and
send the proper acct start packet.

    Should the module register its handlers using the 
switch_channel_add_state_handler() function instead?
And if yes, how could the module pass the channel as a parameter to the 
function since channels
are created and destroyed dynamically (and the module when initialized 
does not have that info).

    I would greatly appreciate your help in pinpointing a proper way to 
call my event handling
routines on a per call leg basis.


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 



From janvb at live.com  Fri Jan 30 04:42:12 2009
From: janvb at live.com (Jan Berger)
Date: Fri, 30 Jan 2009 13:42:12 +0100
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters -
 status
In-Reply-To: <200901301133.30826.hselasky@c2i.net>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<C5A611E3.4CD8D%krice@freeswitch.org>
	<BAY119-W3ABD90DFEA06701EE0292D2C80@phx.gbl> 
	<200901301133.30826.hselasky@c2i.net>
Message-ID: <BLU127-W34E767A7A343F3B1F5CDEFD2C60@phx.gbl>


What is the "Optimized Kernel Mechanism"  you reffer to?
 
Jan> From: hselasky at c2i.net> To: freeswitch-dev at lists.freeswitch.org> Date: Fri, 30 Jan 2009 11:33:29 +0100> Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters - status> > Hi,> > I'm currently working on ISDN (E1/T1/BRI) support, D-channel protocol wise and > hardware wise in connection to FreeSwitch. I'm also adding support for analog > phone adapters through the same API. I promised to Mike Jerris several times > I would do something, but then the time ran away ... This week I put down > several hours integrating the ISDN support in FreeSwitch with the ISDN > support in ISDN4BSD, which should end up with a unified BSD-licensed OpenZap > code base.> > If you are interested in giving some feedback you can check out the following > files:> > svn --username anonsvn --password anonsvn \> checkout svn://svn.turbocat.net/i4b/trunk/openzap.hps> > The files that are there are mainly for inter-process communication, and have > been constructed in a way that allows easy porting to other platforms. In > general sockets can be used for this on other UNIX compatible systems as a > fallback mechanism where the optimised kernel mechanism is not present.> > The core (FreeBSD kernel module): Handles routing and broadcasting.> The client (FreeBSD library): Dumb node sending and receiving messages.> > What I plan next is to implement a set of nodes, like [ISDN] controller- > Q921- and Q931- node. Nodes have pre-defined address ranges which they are > listening to. Hence I already have a DSS1 stack, splitting it up into using > message based communication should not be a very big task.> > Then you have the application, like FreeSwitch being a node aswell picking up > broadcast events, which are mostly incoming calls from Q931 and making > outgoing calls.> > Messages have been split into two categories: Important-frames and > Unimportant-frames. I-Frames gets to use 64 queue entries before getting > dropped. U-Frames gets to use only 32 queue entries before getting dropped. > These numbers have not been tuned yet.> > Also I have thought through message timing. In bigger systems I see that it is > required to dither messages in time. On cannot simply receive 30x400 bytes on > an E1 and burst it into the system, because the message queues will overflow > pretty quickly. Instead the interrupt rate of the hardware needs to be > increased so that 400 bytes are received in "1/30 * interval" fashion. This > will generally improve the system and load-balance in time.> > > The generic Message Header looks like this:> > /* NOTE: All fields are little endian */> > struct zap_hdr {> uint8_t dwDstNode[4]; /* Destination Node */> uint8_t dwSrcNode[4]; /* Source Node */> uint8_t wCommand[2]; /* Command Number */> uint8_t wMsgNum[2]; /* Message Number */> uint8_t bReserved[4]; /* Reserved bytes */> };> > If you have access to ISDN testing equipment and would like to help test the > non-EuroISDN variants of my upcoming ISDN implementation, please let me know.> > I expect to have something up and running within the next month, hopefully not > running out of time this time :-)> > --HPS> > _______________________________________________> Freeswitch-dev mailing list> Freeswitch-dev at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev> http://www.freeswitch.org
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From hselasky at c2i.net  Fri Jan 30 05:20:16 2009
From: hselasky at c2i.net (Hans Petter Selasky)
Date: Fri, 30 Jan 2009 14:20:16 +0100
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters -
	status
In-Reply-To: <BLU127-W34E767A7A343F3B1F5CDEFD2C60@phx.gbl>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<200901301133.30826.hselasky@c2i.net>
	<BLU127-W34E767A7A343F3B1F5CDEFD2C60@phx.gbl>
Message-ID: <200901301420.16812.hselasky@c2i.net>

On Friday 30 January 2009, Jan Berger wrote:
> What is the "Optimized Kernel Mechanism"  you reffer to?
>

Hi Jan,

It is a set of system calls or IOCTLs used to transfer messages between the 
kernel and userland application. Currently copyin/copyout is used, and only 
one message is transferred at a time, though the possibility is there to 
change it.

What I want to acchieve is to reduce the number of system calls needed if 
there are multiple messages to be received or transmitted at a time, by 
transferring multiple messages in a single syscall. When using read()/write() 
on a socket this is not possible. You get all the data back to back in one 
buffer, instead of in separate buffers that can be separately freed.

--HPS


From regs at kinetix.gr  Fri Jan 30 05:48:21 2009
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 30 Jan 2009 15:48:21 +0200
Subject: [Freeswitch-dev] getting a session pointer within an event handler
Message-ID: <49830525.7060307@kinetix.gr>


How can I get a session pointer within an event handler?

I tried to extract the uuid from the event header and then I tried to 
locate the session using the uuid :

static void my_event_handler(switch_event_t *event)
{

         switch_assert(event);

         switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, 
"channel name : %s\n",switch_event_get_header_nil(event, "channel-name"));

        char* uuid = switch_event_get_header(event, "session-id");

        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "uuid : 
%s\n",uuid);

        switch_core_session_t *session;

        session = switch_core_session_locate(uuid);

        switch_assert(session);

    ...

}

But the uuid is always null. The events that I am binding my routine is 
for CHANNEL_CREATE (I tried CHANNEL_EXECUTE too, same result)

-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 



From anthony.minessale at gmail.com  Fri Jan 30 05:58:26 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Jan 2009 07:58:26 -0600
Subject: [Freeswitch-dev] getting a session pointer within an event
	handler
In-Reply-To: <49830525.7060307@kinetix.gr>
References: <49830525.7060307@kinetix.gr>
Message-ID: <191c3a030901300558o2df83250h59cae52dbc97a562@mail.gmail.com>

The name of the header is.

Unique-ID

On Fri, Jan 30, 2009 at 7:48 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>
> How can I get a session pointer within an event handler?
>
> I tried to extract the uuid from the event header and then I tried to
> locate the session using the uuid :
>
> static void my_event_handler(switch_event_t *event)
> {
>
>         switch_assert(event);
>
>         switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
> "channel name : %s\n",switch_event_get_header_nil(event, "channel-name"));
>
>        char* uuid = switch_event_get_header(event, "session-id");
>
>        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "uuid :
> %s\n",uuid);
>
>        switch_core_session_t *session;
>
>        session = switch_core_session_locate(uuid);
>
>        switch_assert(session);
>
>    ...
>
> }
>
> But the uuid is always null. The events that I am binding my routine is
> for CHANNEL_CREATE (I tried CHANNEL_EXECUTE too, same result)
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Jan 30 06:10:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Jan 2009 08:10:04 -0600
Subject: [Freeswitch-dev] mod_radius_cdr behavior and questions
In-Reply-To: <4982E0FF.8070805@kinetix.gr>
References: <4982E0FF.8070805@kinetix.gr>
Message-ID: <191c3a030901300610v238feaf6nfcf79203811eb8b8@mail.gmail.com>

use a queue object to send the data in a dynamic struct to the other thread.

1) create a global queue.
2) create a struct with all the info you need to send.

on the event handler.

1) malloc a new struct of that type.
2) memset it to all 0.
3) populate the struct.
4) write the data into the queue.

launch a thread at startup that does a blocking wait on the same queue

1) pop the void pointer off the queue.
2) cast it into your struct.
3) extract the data from the struct and send it over radius.
4) destroy the struct with free and loop.


On Fri, Jan 30, 2009 at 5:14 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

> Hi,
>
>    I am tweaking the mod_radius_cdr module to archive the behavior
> that most NASes have (i.e. accounting packets are sent in a separate
> thread so that the submission does not interfere with the execution of
> the call). While doing that, however, I bumped into another behavior of
> the module that I think is not desirable (at least by me) :
>
>    While on a bridge, the module sends one acct start packet at the
> beginning of the originating
> leg (on_routing event handler) and two acct stop packets at the end of
> each leg
> (inbound and outbound). My opinion is that it should send one accounting
> start
> packet at the beginning of each call leg (inbound, outbound) resulting
> to a total
> number of two acct start packets. It is generally accepted that acct
> start/stop packets
> come in pairs so that billing applications can handle them accordingly.
>
>    Some NAS's radius radius implementations have some other
> configuration modes
> like the Cisco's RADIUS Packet Suppression. When in this mode the Cisco NAS
> sends only an accounting start/stop pair at the end of a final dialpeer
> attempt (and suppresses
> all the previous failed dialpeer attempts) thus resulting to less
> network traffic. Other
> NASes (such as MERA MVTS) can send a start/stop pair for each leg OR a
> start/stop
> pair for each end to end call, depending on the configuration. Opensips
> follows
> the star/stop pair by call leg paradigm. No matter what the
> implementation, all of them
> always send a acct start/stop pair. This is a common thing. And all the
> billing platforms
> can deal only with paired start/stops.
>
>    The current module behavior (one start two stops) can complicate
> things since the
> radius server would not know how to match the second stop packet with
> its equivalent start.
>
>    Before I get the infamous answer "pathes welcomed" I would like to
> state that I am
> already involved in changing this behavior (through a patch). So my real
> question is this :
>
> I noticed that the module uses the switch_core_add_state_handler
> function to register
> its handler table :
>
> switch_core_add_state_handler(&state_handlers);
>
> So the on_routing ( or the on_execute) event happens only when the
> inbound call is started.
> When the outbound call is initiated no handler is available to hook up a
> function and
> send the proper acct start packet.
>
>    Should the module register its handlers using the
> switch_channel_add_state_handler() function instead?
> And if yes, how could the module pass the channel as a parameter to the
> function since channels
> are created and destroyed dynamically (and the module when initialized
> does not have that info).
>
>    I would greatly appreciate your help in pinpointing a proper way to
> call my event handling
> routines on a per call leg basis.
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From regs at kinetix.gr  Fri Jan 30 06:24:29 2009
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 30 Jan 2009 16:24:29 +0200
Subject: [Freeswitch-dev] getting a session pointer within an
	event	handler
In-Reply-To: <191c3a030901300558o2df83250h59cae52dbc97a562@mail.gmail.com>
References: <49830525.7060307@kinetix.gr>
	<191c3a030901300558o2df83250h59cae52dbc97a562@mail.gmail.com>
Message-ID: <49830D9D.9050108@kinetix.gr>

Thanks! I copied the lower-case version from another module while
searching the code. I wonder how that module works (?!) (mod_event_socket.c)

Anthony Minessale wrote:
> The name of the header is.
>
> Unique-ID
>
> On Fri, Jan 30, 2009 at 7:48 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>
>     How can I get a session pointer within an event handler?
>
>     I tried to extract the uuid from the event header and then I tried to
>     locate the session using the uuid :
>
>     static void my_event_handler(switch_event_t *event)
>     {
>
>             switch_assert(event);
>
>             switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
>     "channel name : %s\n",switch_event_get_header_nil(event,
>     "channel-name"));
>
>            char* uuid = switch_event_get_header(event, "session-id");
>
>            switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "uuid :
>     %s\n",uuid);
>
>            switch_core_session_t *session;
>
>            session = switch_core_session_locate(uuid);
>
>            switch_assert(session);
>
>        ...
>
>     }
>
>     But the uuid is always null. The events that I am binding my
>     routine is
>     for CHANNEL_CREATE (I tried CHANNEL_EXECUTE too, same result)
>
>     --
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     -------------------------------------------
>
>
>     _______________________________________________
>     Freeswitch-dev mailing list
>     Freeswitch-dev at lists.freeswitch.org
>     <mailto:Freeswitch-dev at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From regs at kinetix.gr  Fri Jan 30 06:27:49 2009
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 30 Jan 2009 16:27:49 +0200
Subject: [Freeswitch-dev] mod_radius_cdr behavior and questions
In-Reply-To: <191c3a030901300610v238feaf6nfcf79203811eb8b8@mail.gmail.com>
References: <4982E0FF.8070805@kinetix.gr>
	<191c3a030901300610v238feaf6nfcf79203811eb8b8@mail.gmail.com>
Message-ID: <49830E65.9060106@kinetix.gr>

Wow. I didn't expect so much detailing :)

Thanks for the idea.

My implementation is different though, but yours seems to be better.

I will conclude what I started doing now and get back to you with the 
results.

If something is wrong against my implementation I will try doing it your 
way.

Thanks again!

Anthony Minessale wrote:
> use a queue object to send the data in a dynamic struct to the other 
> thread.
>
> 1) create a global queue.
> 2) create a struct with all the info you need to send.
>
> on the event handler.
>
> 1) malloc a new struct of that type.
> 2) memset it to all 0.
> 3) populate the struct.
> 4) write the data into the queue.
>
> launch a thread at startup that does a blocking wait on the same queue
>
> 1) pop the void pointer off the queue.
> 2) cast it into your struct.
> 3) extract the data from the struct and send it over radius.
> 4) destroy the struct with free and loop.
>
>
> On Fri, Jan 30, 2009 at 5:14 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     Hi,
>
>        I am tweaking the mod_radius_cdr module to archive the behavior
>     that most NASes have (i.e. accounting packets are sent in a separate
>     thread so that the submission does not interfere with the execution of
>     the call). While doing that, however, I bumped into another
>     behavior of
>     the module that I think is not desirable (at least by me) :
>
>        While on a bridge, the module sends one acct start packet at the
>     beginning of the originating
>     leg (on_routing event handler) and two acct stop packets at the end of
>     each leg
>     (inbound and outbound). My opinion is that it should send one
>     accounting
>     start
>     packet at the beginning of each call leg (inbound, outbound) resulting
>     to a total
>     number of two acct start packets. It is generally accepted that acct
>     start/stop packets
>     come in pairs so that billing applications can handle them
>     accordingly.
>
>        Some NAS's radius radius implementations have some other
>     configuration modes
>     like the Cisco's RADIUS Packet Suppression. When in this mode the
>     Cisco NAS
>     sends only an accounting start/stop pair at the end of a final
>     dialpeer
>     attempt (and suppresses
>     all the previous failed dialpeer attempts) thus resulting to less
>     network traffic. Other
>     NASes (such as MERA MVTS) can send a start/stop pair for each leg OR a
>     start/stop
>     pair for each end to end call, depending on the configuration.
>     Opensips
>     follows
>     the star/stop pair by call leg paradigm. No matter what the
>     implementation, all of them
>     always send a acct start/stop pair. This is a common thing. And
>     all the
>     billing platforms
>     can deal only with paired start/stops.
>
>        The current module behavior (one start two stops) can complicate
>     things since the
>     radius server would not know how to match the second stop packet with
>     its equivalent start.
>
>        Before I get the infamous answer "pathes welcomed" I would like to
>     state that I am
>     already involved in changing this behavior (through a patch). So
>     my real
>     question is this :
>
>     I noticed that the module uses the switch_core_add_state_handler
>     function to register
>     its handler table :
>
>     switch_core_add_state_handler(&state_handlers);
>
>     So the on_routing ( or the on_execute) event happens only when the
>     inbound call is started.
>     When the outbound call is initiated no handler is available to
>     hook up a
>     function and
>     send the proper acct start packet.
>
>        Should the module register its handlers using the
>     switch_channel_add_state_handler() function instead?
>     And if yes, how could the module pass the channel as a parameter
>     to the
>     function since channels
>     are created and destroyed dynamically (and the module when initialized
>     does not have that info).
>
>        I would greatly appreciate your help in pinpointing a proper way to
>     call my event handling
>     routines on a per call leg basis.
>
>
>     --
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     -------------------------------------------
>
>
>     _______________________________________________
>     Freeswitch-dev mailing list
>     Freeswitch-dev at lists.freeswitch.org
>     <mailto:Freeswitch-dev at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From rob.charlton at savageminds.com  Fri Jan 30 07:28:34 2009
From: rob.charlton at savageminds.com (Rob Charlton)
Date: Fri, 30 Jan 2009 15:28:34 +0000
Subject: [Freeswitch-dev] getting a session pointer within
	an	event	handler
In-Reply-To: <49830D9D.9050108@kinetix.gr>
References: <49830525.7060307@kinetix.gr>	<191c3a030901300558o2df83250h59cae52dbc97a562@mail.gmail.com>
	<49830D9D.9050108@kinetix.gr>
Message-ID: <49831CA2.1010200@savageminds.com>

Apostolos Pantsiopoulos wrote:
> Thanks! I copied the lower-case version from another module while
> searching the code. I wonder how that module works (?!) 
> (mod_event_socket.c)
>
Because switch_event_get_header uses strcasecmp so it is case 
insensitive. The important thing is to use unique-id rather than session-id.

Rob

-- 
Rob Charlton
Savage Minds Ltd



From janvb at live.com  Fri Jan 30 07:30:21 2009
From: janvb at live.com (Jan Berger)
Date: Fri, 30 Jan 2009 16:30:21 +0100
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters -
 status
In-Reply-To: <200901301133.30826.hselasky@c2i.net>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<C5A611E3.4CD8D%krice@freeswitch.org>
	<BAY119-W3ABD90DFEA06701EE0292D2C80@phx.gbl> 
	<200901301133.30826.hselasky@c2i.net>
Message-ID: <BLU127-W4650A4452910C3B76FB608D2C60@phx.gbl>


hi Hans,
 
The standard for a PABX on G.711 is 6ms packaging of data - this is 48 byte per channel. If you go higher than this you start getting to much latency, if you go lower than this you start using to much CPU.
 
400 byte is 50ms, meaning we have spent a lot of the latency budget already here, but this shoud be run-time configurable per channel as per need.
 
- G.711 6 ms
- IVR is 250 ms
- 729 10 ms
- 723 30 ms
 
etc.
 
You need to take into consideration that you are on a system with 8, 16 or 32 E1's of whick some is running IVR, others SIP, others swicthing back out on a different E1 etc.
 
Jan
 
> > Also I have thought through message timing. In bigger systems I see that it is > required to dither messages in time. On cannot simply receive 30x400 bytes on > an E1 and burst it into the system, because the message queues will overflow > pretty quickly. Instead the interrupt rate of the hardware needs to be > increased so that 400 bytes are received in "1/30 * interval" fashion. This > will generally improve the system and load-balance in time.> > > The generic Message Header looks like this:> > /* NOTE: All fields are little endian */> > struct zap_hdr {> uint8_t dwDstNode[4]; /* Destination Node */> uint8_t dwSrcNode[4]; /* Source Node */> uint8_t wCommand[2]; /* Command Number */> uint8_t wMsgNum[2]; /* Message Number */> uint8_t bReserved[4]; /* Reserved bytes */> };> > If you have access to ISDN testing equipment and would like to help test the > non-EuroISDN variants of my upcoming ISDN implementation, please let me know.> > I expect to have something up and running within the next month, hopefully not > running out of time this time :-)> > --HPS> > _______________________________________________> Freeswitch-dev mailing list> Freeswitch-dev at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev> http://www.freeswitch.org
_________________________________________________________________
News, entertainment and everything you care about at Live.com. Get it now!
http://www.live.com/getstarted.aspx
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From brian at freeswitch.org  Fri Jan 30 07:35:44 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 30 Jan 2009 09:35:44 -0600
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters -
	status
In-Reply-To: <BLU127-W4650A4452910C3B76FB608D2C60@phx.gbl>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<C5A611E3.4CD8D%krice@freeswitch.org>
	<BAY119-W3ABD90DFEA06701EE0292D2C80@phx.gbl>
	<200901301133.30826.hselasky@c2i.net>
	<BLU127-W4650A4452910C3B76FB608D2C60@phx.gbl>
Message-ID: <082B044A-3FD7-4495-882A-58EB3B9C7057@freeswitch.org>

Can you elaborate on this G.711 at 6ms?  I have never seen this odd  
number.

/b

On Jan 30, 2009, at 9:30 AM, Jan Berger wrote:

> hi Hans,
>
> The standard for a PABX on G.711 is 6ms packaging of data - this is  
> 48 byte per channel. If you go higher than this you start getting to  
> much latency, if you go lower than this you start using to much CPU.
>
> 400 byte is 50ms, meaning we have spent a lot of the latency budget  
> already here, but this shoud be run-time configurable per channel as  
> per need.
>
> - G.711 6 ms
> - IVR is 250 ms
> - 729 10 ms
> - 723 30 ms
>
> etc.
>
> You need to take into consideration that you are on a system with 8,  
> 16 or 32 E1's of whick some is running IVR, others SIP, others  
> swicthing back out on a different E1 etc.
>
> Jan
>

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From janvb at live.com  Fri Jan 30 08:03:01 2009
From: janvb at live.com (Jan Berger)
Date: Fri, 30 Jan 2009 17:03:01 +0100
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters
	-	status
In-Reply-To: <082B044A-3FD7-4495-882A-58EB3B9C7057@freeswitch.org>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<C5A611E3.4CD8D%krice@freeswitch.org>
	<BAY119-W3ABD90DFEA06701EE0292D2C80@phx.gbl>
	<200901301133.30826.hselasky@c2i.net>
	<BLU127-W4650A4452910C3B76FB608D2C60@phx.gbl> 
	<082B044A-3FD7-4495-882A-58EB3B9C7057@freeswitch.org>
Message-ID: <BAY122-W408FC6E9B3FA67077015BD2C60@phx.gbl>


It's an old hand-rule, nothing more.
 
The ideal situation would be 1 byte - 0.125 ms latency in switching, but this means a computer masturbating at 8000 interrupt a second - and can you imagine what this will do with your CPU?
 
Write a small test application that perform memcpy operations and calculate the transer rate.
 
When test on 1 byte, 5 bytes, 10 bytes 25 bytes 50 bytes
 
What you will see is that the transfer rate increase with the size of the packet - 50 byte is ca 50 byte faster than 1 byte in x-fer.
 
Now test with 100 bytes 250 bytes 500 bytes 1000 bytes etc
 
And you will see that little has changed from ca 50 bytes in xfer rate.
 
48 byte basicaly is a good trade between latency and cpu usage. The test might vary dependent on processor , but....
 
---
 
10 ms is however more or less the same, and might be a better number today. It goes for 711, 729 and 723 ...
 
Jan



From: brian at freeswitch.orgTo: freeswitch-dev at lists.freeswitch.orgDate: Fri, 30 Jan 2009 09:35:44 -0600Subject: Re: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters - statusCan you elaborate on this G.711 at 6ms?  I have never seen this odd number.

/b


On Jan 30, 2009, at 9:30 AM, Jan Berger wrote:
hi Hans, The standard for a PABX on G.711 is 6ms packaging of data - this is 48 byte per channel. If you go higher than this you start getting to much latency, if you go lower than this you start using to much CPU. 400 byte is 50ms, meaning we have spent a lot of the latency budget already here, but this shoud be run-time configurable per channel as per need. - G.711 6 ms- IVR is 250 ms- 729 10 ms- 723 30 ms etc. You need to take into consideration that you are on a system with 8, 16 or 32 E1's of whick some is running IVR, others SIP, others swicthing back out on a different E1 etc. Jan 
_________________________________________________________________
News, entertainment and everything you care about at Live.com. Get it now!
http://www.live.com/getstarted.aspx
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From anthony.minessale at gmail.com  Fri Jan 30 08:11:20 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Jan 2009 10:11:20 -0600
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters -
	status
In-Reply-To: <BAY122-W408FC6E9B3FA67077015BD2C60@phx.gbl>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<C5A611E3.4CD8D%krice@freeswitch.org>
	<BAY119-W3ABD90DFEA06701EE0292D2C80@phx.gbl>
	<200901301133.30826.hselasky@c2i.net>
	<BLU127-W4650A4452910C3B76FB608D2C60@phx.gbl>
	<082B044A-3FD7-4495-882A-58EB3B9C7057@freeswitch.org>
	<BAY122-W408FC6E9B3FA67077015BD2C60@phx.gbl>
Message-ID: <191c3a030901300811j1ffa44f0xdc8eb66e9b0b88e@mail.gmail.com>

That is one of the benefits of sangoma hardware.
They have an mtu value in the driver you can set to 80 (10ms) which despite
the interval the channel is using
it will operate at 10ms interrupts.  Digium's zaptel, umm i mean dahdi uses
the aforementioned masturbation technique and any hardware that interfaces
to it must comply.  Since most of use do not want to be forced to masturbate
with dahdi we are seeking an alternative, hence OpenZAP

OpenZAP so far is only a user land interface with plug-ins for i/o and
signaling.
Eventually the goal is to make a cross-platform kernel layer interface as
well that can abstract TDM and RTP hardware alike and allow the creation of
handles that can be cross-connected inside the kernel as well as in and out
of userland depending on the needs.




On Fri, Jan 30, 2009 at 10:03 AM, Jan Berger <janvb at live.com> wrote:

>  It's an old hand-rule, nothing more.
>
> The ideal situation would be 1 byte - 0.125 ms latency in switching, but
> this means a computer masturbating at 8000 interrupt a second - and can you
> imagine what this will do with your CPU?
>
> Write a small test application that perform memcpy operations and calculate
> the transer rate.
>
> When test on 1 byte, 5 bytes, 10 bytes 25 bytes 50 bytes
>
> What you will see is that the transfer rate increase with the size of the
> packet - 50 byte is ca 50 byte faster than 1 byte in x-fer.
>
> Now test with 100 bytes 250 bytes 500 bytes 1000 bytes etc
>
> And you will see that little has changed from ca 50 bytes in xfer rate.
>
> 48 byte basicaly is a good trade between latency and cpu usage. The test
> might vary dependent on processor , but....
>
> ---
>
> 10 ms is however more or less the same, and might be a better number today.
> It goes for 711, 729 and 723 ...
>
> Jan
>
>
> ------------------------------
>
> From: brian at freeswitch.org
> To: freeswitch-dev at lists.freeswitch.org
> Date: Fri, 30 Jan 2009 09:35:44 -0600
> Subject: Re: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters -
> status
>
>
> Can you elaborate on this G.711 at 6ms?  I have never seen this odd number.
>
> /b
>
>  On Jan 30, 2009, at 9:30 AM, Jan Berger wrote:
>
> hi Hans,
>
> The standard for a PABX on G.711 is 6ms packaging of data - this is 48 byte
> per channel. If you go higher than this you start getting to much latency,
> if you go lower than this you start using to much CPU.
>
> 400 byte is 50ms, meaning we have spent a lot of the latency budget already
> here, but this shoud be run-time configurable per channel as per need.
>
> - G.711 6 ms
> - IVR is 250 ms
> - 729 10 ms
> - 723 30 ms
>
> etc.
>
> You need to take into consideration that you are on a system with 8, 16 or
> 32 E1's of whick some is running IVR, others SIP, others swicthing back out
> on a different E1 etc.
>
> Jan
>
>
>
>
> ------------------------------
> Get news, entertainment and everything you care about at Live.com. Check
> it out! <http://www.live.com/getstarted.aspx>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Jan 30 08:14:21 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Jan 2009 10:14:21 -0600
Subject: [Freeswitch-dev] mod_radius_cdr behavior and questions
In-Reply-To: <49830E65.9060106@kinetix.gr>
References: <4982E0FF.8070805@kinetix.gr>
	<191c3a030901300610v238feaf6nfcf79203811eb8b8@mail.gmail.com>
	<49830E65.9060106@kinetix.gr>
Message-ID: <191c3a030901300814u4d6315dbk752cd059b38c98b3@mail.gmail.com>

No problem, I have the advantage That I have implemented this technique all
over the place ;)
Your event handler is the recipient end of that same algorithm.

In fact the events are a very good thing to pass into queues.

You could clone the event and insert the clone into the queue and when you
pop it from the
backend thread you can just destroy it there then.



On Fri, Jan 30, 2009 at 8:27 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>  Wow. I didn't expect so much detailing :)
>
> Thanks for the idea.
>
> My implementation is different though, but yours seems to be better.
>
> I will conclude what I started doing now and get back to you with the
> results.
>
> If something is wrong against my implementation I will try doing it your
> way.
>
> Thanks again!
>
> Anthony Minessale wrote:
>
> use a queue object to send the data in a dynamic struct to the other
> thread.
>
> 1) create a global queue.
> 2) create a struct with all the info you need to send.
>
> on the event handler.
>
> 1) malloc a new struct of that type.
> 2) memset it to all 0.
> 3) populate the struct.
> 4) write the data into the queue.
>
> launch a thread at startup that does a blocking wait on the same queue
>
> 1) pop the void pointer off the queue.
> 2) cast it into your struct.
> 3) extract the data from the struct and send it over radius.
> 4) destroy the struct with free and loop.
>
>
> On Fri, Jan 30, 2009 at 5:14 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:
>
>> Hi,
>>
>>    I am tweaking the mod_radius_cdr module to archive the behavior
>> that most NASes have (i.e. accounting packets are sent in a separate
>> thread so that the submission does not interfere with the execution of
>> the call). While doing that, however, I bumped into another behavior of
>> the module that I think is not desirable (at least by me) :
>>
>>    While on a bridge, the module sends one acct start packet at the
>> beginning of the originating
>> leg (on_routing event handler) and two acct stop packets at the end of
>> each leg
>> (inbound and outbound). My opinion is that it should send one accounting
>> start
>> packet at the beginning of each call leg (inbound, outbound) resulting
>> to a total
>> number of two acct start packets. It is generally accepted that acct
>> start/stop packets
>> come in pairs so that billing applications can handle them accordingly.
>>
>>    Some NAS's radius radius implementations have some other
>> configuration modes
>> like the Cisco's RADIUS Packet Suppression. When in this mode the Cisco
>> NAS
>> sends only an accounting start/stop pair at the end of a final dialpeer
>> attempt (and suppresses
>> all the previous failed dialpeer attempts) thus resulting to less
>> network traffic. Other
>> NASes (such as MERA MVTS) can send a start/stop pair for each leg OR a
>> start/stop
>> pair for each end to end call, depending on the configuration. Opensips
>> follows
>> the star/stop pair by call leg paradigm. No matter what the
>> implementation, all of them
>> always send a acct start/stop pair. This is a common thing. And all the
>> billing platforms
>> can deal only with paired start/stops.
>>
>>    The current module behavior (one start two stops) can complicate
>> things since the
>> radius server would not know how to match the second stop packet with
>> its equivalent start.
>>
>>    Before I get the infamous answer "pathes welcomed" I would like to
>> state that I am
>> already involved in changing this behavior (through a patch). So my real
>> question is this :
>>
>> I noticed that the module uses the switch_core_add_state_handler
>> function to register
>> its handler table :
>>
>> switch_core_add_state_handler(&state_handlers);
>>
>> So the on_routing ( or the on_execute) event happens only when the
>> inbound call is started.
>> When the outbound call is initiated no handler is available to hook up a
>> function and
>> send the proper acct start packet.
>>
>>    Should the module register its handlers using the
>> switch_channel_add_state_handler() function instead?
>> And if yes, how could the module pass the channel as a parameter to the
>> function since channels
>> are created and destroyed dynamically (and the module when initialized
>> does not have that info).
>>
>>    I would greatly appreciate your help in pinpointing a proper way to
>> call my event handling
>> routines on a per call leg basis.
>>
>>
>> --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-dev mailing listFreeswitch-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From mike at jerris.com  Fri Jan 30 08:17:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 30 Jan 2009 11:17:27 -0500
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters -
	status
In-Reply-To: <191c3a030901300811j1ffa44f0xdc8eb66e9b0b88e@mail.gmail.com>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<C5A611E3.4CD8D%krice@freeswitch.org>
	<BAY119-W3ABD90DFEA06701EE0292D2C80@phx.gbl>
	<200901301133.30826.hselasky@c2i.net>
	<BLU127-W4650A4452910C3B76FB608D2C60@phx.gbl>
	<082B044A-3FD7-4495-882A-58EB3B9C7057@freeswitch.org>
	<BAY122-W408FC6E9B3FA67077015BD2C60@phx.gbl>
	<191c3a030901300811j1ffa44f0xdc8eb66e9b0b88e@mail.gmail.com>
Message-ID: <F7D97B80-26DF-4870-B7B8-9E7C70397188@jerris.com>

Another useful advantage without the latency tradeoff is to read a  
full span or card every 1ms  (or whatever sane interval) and chop up  
the bytes in userland.  The latency argument dies if you are bridging  
to voip, so if you can handle bridging the tdm to tdm channels in  
kernel space, you could easily get away with 10ms reads to user space  
as well.

Mike



On Jan 30, 2009, at 11:11 AM, Anthony Minessale wrote:

> That is one of the benefits of sangoma hardware.
> They have an mtu value in the driver you can set to 80 (10ms) which  
> despite the interval the channel is using
> it will operate at 10ms interrupts.  Digium's zaptel, umm i mean  
> dahdi uses the aforementioned masturbation technique and any  
> hardware that interfaces to it must comply.  Since most of use do  
> not want to be forced to masturbate with dahdi we are seeking an  
> alternative, hence OpenZAP
>
> OpenZAP so far is only a user land interface with plug-ins for i/o  
> and signaling.
> Eventually the goal is to make a cross-platform kernel layer  
> interface as well that can abstract TDM and RTP hardware alike and  
> allow the creation of handles that can be cross-connected inside the  
> kernel as well as in and out of userland depending on the needs.
>
>
>
>
> On Fri, Jan 30, 2009 at 10:03 AM, Jan Berger <janvb at live.com> wrote:
> It's an old hand-rule, nothing more.
>
> The ideal situation would be 1 byte - 0.125 ms latency in switching,  
> but this means a computer masturbating at 8000 interrupt a second -  
> and can you imagine what this will do with your CPU?
>
> Write a small test application that perform memcpy operations and  
> calculate the transer rate.
>
> When test on 1 byte, 5 bytes, 10 bytes 25 bytes 50 bytes
>
> What you will see is that the transfer rate increase with the size  
> of the packet - 50 byte is ca 50 byte faster than 1 byte in x-fer.
>
> Now test with 100 bytes 250 bytes 500 bytes 1000 bytes etc
>
> And you will see that little has changed from ca 50 bytes in xfer  
> rate.
>
> 48 byte basicaly is a good trade between latency and cpu usage. The  
> test might vary dependent on processor , but....
>
> ---
>
> 10 ms is however more or less the same, and might be a better number  
> today. It goes for 711, 729 and 723 ...
>
> Jan
>
>
>
> From: brian at freeswitch.org
>
> To: freeswitch-dev at lists.freeswitch.org
> Date: Fri, 30 Jan 2009 09:35:44 -0600
> Subject: Re: [Freeswitch-dev] FreeSwitch + ISDN + analog phone  
> adapters - status
>
>
> Can you elaborate on this G.711 at 6ms?  I have never seen this odd  
> number.
>
> /b
>
> On Jan 30, 2009, at 9:30 AM, Jan Berger wrote:
>
> hi Hans,
>
> The standard for a PABX on G.711 is 6ms packaging of data - this is  
> 48 byte per channel. If you go higher than this you start getting to  
> much latency, if you go lower than this you start using to much CPU.
>
> 400 byte is 50ms, meaning we have spent a lot of the latency budget  
> already here, but this shoud be run-time configurable per channel as  
> per need.
>
> - G.711 6 ms
> - IVR is 250 ms
> - 729 10 ms
> - 723 30 ms
>
> etc.
>
> You need to take into consideration that you are on a system with 8,  
> 16 or 32 E1's of whick some is running IVR, others SIP, others  
> swicthing back out on a different E1 etc.
>
> Jan
>
>
>
> Get news, entertainment and everything you care about at Live.com.  
> Check it out!
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
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> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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From anthony.minessale at gmail.com  Fri Jan 30 08:40:30 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Jan 2009 10:40:30 -0600
Subject: [Freeswitch-dev] freeswitch development question (memory
	allocation?) [SOLVED]
In-Reply-To: <4982B406.1050107@kinetix.gr>
References: <4981A3AB.2000803@kinetix.gr> <4981A876.5030605@kinetix.gr>
	<2EDFD0BF-7767-42C2-8B09-48A75B0B2F08@jerris.com>
	<4981ADCF.6050802@kinetix.gr>
	<191c3a030901290555i58186cc8p2cdecd7b700f33dc@mail.gmail.com>
	<4981C7A0.7090000@kinetix.gr> <4982B406.1050107@kinetix.gr>
Message-ID: <191c3a030901300840p43ec2a95x69f2b50e4f1b2ee3@mail.gmail.com>

You cannot detain the session at all or you are defeating the purpose of
doing it in another thread.
I already gave you the complete instructions on how to do it properly so
that's all I have to offer.

If you are fetching the session from the hangup event you also run the risk
for a race where the session may have exited already and you will miss it.

Again, you have the channel_hangup event complete with all the data you
would ever need to do the cdr, just clone the event and stick it into a
queue.


On Fri, Jan 30, 2009 at 2:02 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>  Hi,
>
> One more clarification please :
>
> In order for my thread to use the session data before the session is
> destroyed I am using
> my own mutex and condition signal. e.g.
>
> (pseudo code follows) :
>
> thread_function(switch_thread_t *thread, void *ptr){
>
> // the pointer ptr is a struct that contains a pointer to session,the mutex
> and condition created in the calling thread
>
> (session dependend stuff)
>
> switch_thread_cond_signal(sth->cond); //sends signal to calling thread that
> we stopped using the session so it can exit
>
> (session independent stuff)
>
> }
> calling_function(switch_core_session_t *session){
>
>        (init stuff)
>
>         switch_mutex_init(&thread_params->mutex, SWITCH_MUTEX_NESTED,
> pool);
>         switch_thread_cond_create(&thread_params->cond, pool);
>
>        (thread creation stuff)
>
>         switch_mutex_lock( thread_params->mutex );
>
>         switch_thread_cond_wait(thread_params->cond, thread_params->mutex);
> // this is were the calling thread waits for a signal
>
>         switch_mutex_unlock( thread_params->mutex );
>
>         return SWITCH_STATUS_SUCCESS;
>
> }
>
> Is the above correct? Is there another way to create a "wait" state for the
> calling thread until the child thread
> is done with the session data?
>
>
>
>
> Apostolos Pantsiopoulos wrote:
>
> Thanks for the explanations (both Anthony and Michael).
> You both clarified the subject quite well.
>
> Anthony Minessale wrote:
>
> if your intent is to keep the data around longer than the session in it's
> own thread, you will want to make a new pool
> and use that pool to launch the thread.  pass the pool into the thread as a
> member of your thread private data and free
> the pool before you exit the thread.
>
> You cannot bring the session with you into the thread unless you read lock
> the session first so it will not be destroyed while
> you are using it in the thread.
>
> Be aware keeping the session around in the thread just a wild goose chase
> from doing it the why it already was doing it.
> The session thread is already independent and it does not stop any other
> call from proceeding so you may as well do all the work
> in your session thread in the state change handler like it already does and
> just add more strict timeout.
>
>
> On Thu, Jan 29, 2009 at 7:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:
>
>> The question is : which pool should I use for each different thing I am
>> trying to accomplish?
>>
>> For instance the code below is part of my mod_radius_cdr experimentation.
>> When I am making use of a pool within a module should I :
>>
>> a) create a new pool?
>> b) make use of an existing one?
>> c) what kind of pool should I use?
>> d) is it really just one pool (accessed through different handlers) and I
>> am making myself look like a fool so far?
>>
>>
>>
>>
>>
>> Michael Jerris wrote:
>>
>> I am not sure even after re-reading what your question is, could you
>> try to rephrase?
>>
>> Mike
>>
>> On Jan 29, 2009, at 8:00 AM, Apostolos Pantsiopoulos wrote:
>>
>>
>>
>>  I found it out by myself!
>> (why is it that we always come with the solution right after posting
>> to
>> the list?)
>>
>> I inserted :
>>
>> thread_params = switch_core_session_alloc(session,
>> sizeof(*thread_params));
>>
>> before the pool initialization.
>>
>> But still,  can I get some answers to the questions bellow about
>> how to effectively handle memory allocations?
>>
>>
>>
>> Apostolos Pantsiopoulos wrote:
>>
>>
>>  I have the code below :
>>
>> struct radacct_thread_handle {
>>        switch_core_session_t *session;
>>        switch_mutex_t *mutex;
>>        switch_thread_cond_t *cond;
>> };
>>
>> static switch_status_t my_on_routing(switch_core_session_t *session){
>>
>>        switch_thread_t *thread;
>>        switch_threadattr_t *thd_attr = NULL;
>>
>>        switch_memory_pool_t *pool;
>>
>>        struct radacct_thread_handle *thread_params = NULL;
>>
>>        pool = switch_core_session_get_pool(session);
>>
>>        thread_params->session = session;
>>
>>        ...
>>
>> }
>>
>> when the program reaches the last line (thread_params->session =
>> session;)
>> I get a core dump. Is this a memory allocation error? Is it because
>> I am
>> making
>> use of the wrong pool? Please enlighten me because I am not an
>> experienced c
>> programmer, and I am struggling to get familiar with the FS API.
>>
>>
>>
>>
>>  _______________________________________________
>> Freeswitch-dev mailing listFreeswitch-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>>
>>
>>
>>   --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-dev mailing listFreeswitch-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-dev mailing listFreeswitch-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From hselasky at c2i.net  Fri Jan 30 09:11:09 2009
From: hselasky at c2i.net (Hans Petter Selasky)
Date: Fri, 30 Jan 2009 18:11:09 +0100
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters -
	status
In-Reply-To: <F7D97B80-26DF-4870-B7B8-9E7C70397188@jerris.com>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<191c3a030901300811j1ffa44f0xdc8eb66e9b0b88e@mail.gmail.com>
	<F7D97B80-26DF-4870-B7B8-9E7C70397188@jerris.com>
Message-ID: <200901301811.10384.hselasky@c2i.net>

On Friday 30 January 2009, Michael Jerris wrote:
> Another useful advantage without the latency tradeoff is to read a
> full span or card every 1ms  (or whatever sane interval) and chop up
> the bytes in userland.  The latency argument dies if you are bridging
> to voip, so if you can handle bridging the tdm to tdm channels in
> kernel space, you could easily get away with 10ms reads to user space
> as well.
>
> Mike

Hi,

Got your point.

In between:

I see more in Jan's argument. 48x30 bytes = 1440 bytes, which is an 
interesting size for full span E1.

--HPS


From janvb at live.com  Fri Jan 30 14:45:19 2009
From: janvb at live.com (Jan Berger)
Date: Fri, 30 Jan 2009 23:45:19 +0100
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters
	-	status
In-Reply-To: <200901301811.10384.hselasky@c2i.net>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<191c3a030901300811j1ffa44f0xdc8eb66e9b0b88e@mail.gmail.com>
	<F7D97B80-26DF-4870-B7B8-9E7C70397188@jerris.com> 
	<200901301811.10384.hselasky@c2i.net>
Message-ID: <BAY127-W8C49BB174910159A0804ED2C60@phx.gbl>


What do you want to achieve by moving the entire span up in user space unprocessed?
 
Jan> From: hselasky at c2i.net> To: freeswitch-dev at lists.freeswitch.org> Date: Fri, 30 Jan 2009 18:11:09 +0100> Subject: Re: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters - status> > On Friday 30 January 2009, Michael Jerris wrote:> > Another useful advantage without the latency tradeoff is to read a> > full span or card every 1ms (or whatever sane interval) and chop up> > the bytes in userland. The latency argument dies if you are bridging> > to voip, so if you can handle bridging the tdm to tdm channels in> > kernel space, you could easily get away with 10ms reads to user space> > as well.> >> > Mike> > Hi,> > Got your point.> > In between:> > I see more in Jan's argument. 48x30 bytes = 1440 bytes, which is an > interesting size for full span E1.> > --HPS> > _______________________________________________> Freeswitch-dev mailing list> Freeswitch-dev at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev> http://www.freeswitch.org
_________________________________________________________________
Drag n? drop?Get easy photo sharing with Windows Live? Photos.

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From anthony.minessale at gmail.com  Fri Jan 30 15:16:19 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 30 Jan 2009 17:16:19 -0600
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters -
	status
In-Reply-To: <BAY127-W8C49BB174910159A0804ED2C60@phx.gbl>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<191c3a030901300811j1ffa44f0xdc8eb66e9b0b88e@mail.gmail.com>
	<F7D97B80-26DF-4870-B7B8-9E7C70397188@jerris.com>
	<200901301811.10384.hselasky@c2i.net>
	<BAY127-W8C49BB174910159A0804ED2C60@phx.gbl>
Message-ID: <191c3a030901301516x7a30c8c6ydd9c221d0623860a@mail.gmail.com>

I think the goal would be to reduce the number of trips to and from
userspace by doing the multiplexing in userspace to avoid the context
switches of reading every logical channel as a device.


On Fri, Jan 30, 2009 at 4:45 PM, Jan Berger <janvb at live.com> wrote:

>  What do you want to achieve by moving the entire span up in user space
> unprocessed?
>
> Jan
>
> > From: hselasky at c2i.net
> > To: freeswitch-dev at lists.freeswitch.org
> > Date: Fri, 30 Jan 2009 18:11:09 +0100
> > Subject: Re: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters -
> status
> >
> > On Friday 30 January 2009, Michael Jerris wrote:
> > > Another useful advantage without the latency tradeoff is to read a
> > > full span or card every 1ms (or whatever sane interval) and chop up
> > > the bytes in userland. The latency argument dies if you are bridging
> > > to voip, so if you can handle bridging the tdm to tdm channels in
> > > kernel space, you could easily get away with 10ms reads to user space
> > > as well.
> > >
> > > Mike
> >
> > Hi,
> >
> > Got your point.
> >
> > In between:
> >
> > I see more in Jan's argument. 48x30 bytes = 1440 bytes, which is an
> > interesting size for full span E1.
> >
> > --HPS
> >
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
>
>
> ------------------------------
> What can you do with the new Windows Live? Find out<http://www.microsoft.com/windows/windowslive/default.aspx>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From janvb at live.com  Fri Jan 30 17:40:58 2009
From: janvb at live.com (Jan Berger)
Date: Sat, 31 Jan 2009 02:40:58 +0100
Subject: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters
	-	status
In-Reply-To: <191c3a030901301516x7a30c8c6ydd9c221d0623860a@mail.gmail.com>
References: <BAY125-W42F46B61304888CE624147D2C80@phx.gbl>
	<191c3a030901300811j1ffa44f0xdc8eb66e9b0b88e@mail.gmail.com>
	<F7D97B80-26DF-4870-B7B8-9E7C70397188@jerris.com>
	<200901301811.10384.hselasky@c2i.net>
	<BAY127-W8C49BB174910159A0804ED2C60@phx.gbl> 
	<191c3a030901301516x7a30c8c6ydd9c221d0623860a@mail.gmail.com>
Message-ID: <BAY123-W1537AD5C50E955E9B1E66D2C70@phx.gbl>


I would do this very differently.
 
They key to performance is to reduce memcpy and malloc to a minimum. 
 
1. You feed a page from PCI card and swap this over on interrupt/timer - and at this point you copy data to channel buffer or directly to next B channel.
 
The channel buffers can be shared OS memory, meaning that you write from the PCI card in one end and read (directly from buffer) from app in the other end. Also create the channel buffer as a sircular list of packets and you have no malloc involved.
 
You can also use this for messaging between threads and even apps - it is a well known and extremely fast technique. also let the IPC fall back to IP and you could start spreading FreeSWITCH modules across a network.
 
I have some code somewher that does this so I can set up a demo once I return in a weeks time. 
 
Jan
 



Date: Fri, 30 Jan 2009 17:16:19 -0600From: anthony.minessale at gmail.comTo: freeswitch-dev at lists.freeswitch.orgSubject: Re: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters - statusI think the goal would be to reduce the number of trips to and from userspace by doing the multiplexing in userspace to avoid the context switches of reading every logical channel as a device.
On Fri, Jan 30, 2009 at 4:45 PM, Jan Berger <janvb at live.com> wrote:

What do you want to achieve by moving the entire span up in user space unprocessed?
 Jan> From: hselasky at c2i.net
> To: freeswitch-dev at lists.freeswitch.org> Date: Fri, 30 Jan 2009 18:11:09 +0100
> Subject: Re: [Freeswitch-dev] FreeSwitch + ISDN + analog phone adapters - status> 


> On Friday 30 January 2009, Michael Jerris wrote:> > Another useful advantage without the latency tradeoff is to read a> > full span or card every 1ms (or whatever sane interval) and chop up> > the bytes in userland. The latency argument dies if you are bridging> > to voip, so if you can handle bridging the tdm to tdm channels in> > kernel space, you could easily get away with 10ms reads to user space> > as well.> >> > Mike> > Hi,> > Got your point.> > In between:> > I see more in Jan's argument. 48x30 bytes = 1440 bytes, which is an > interesting size for full span E1.> > --HPS> > _______________________________________________> Freeswitch-dev mailing list> Freeswitch-dev at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev> http://www.freeswitch.org

What can you do with the new Windows Live? Find out_______________________________________________Freeswitch-dev mailing listFreeswitch-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-devUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org-- Anthony Minessale IIFreeSWITCH http://www.freeswitch.org/ClueCon http://www.cluecon.com/AIM: anthmMSN:anthony_minessale at hotmail.comGTALK/JABBER/PAYPAL:anthony.minessale at gmail.comIRC: irc.freenode.net #freeswitchFreeSWITCH Developer Conferencesip:888 at conference.freeswitch.orgiax:guest at conference.freeswitch.org/888googletalk:conf+888 at conference.freeswitch.orgpstn:213-799-1400
_________________________________________________________________
More than messages?check out the rest of the Windows Live?.
http://www.microsoft.com/windows/windowslive/
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From msc at freeswitch.org  Sat Jan 31 11:27:24 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 31 Jan 2009 11:27:24 -0800
Subject: [Freeswitch-dev] ANNOUNCEMENT: FreeSWITCH 1.0.3 RC1 Available
Message-ID: <87f2f3b90901311127o24592759q85d9e442b94a4d75@mail.gmail.com>

The FreeSWITCH development team is happy to announce the immediate
availability of FreeSWITCH 1.0.3 RC1. The download is available here:
http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz

More information regarding this release candidate can be found here:
http://freeswitch.org/node/159

Please help us out by downloading and running the latest FreeSWITCH
release. Thank you all for your continued support of this awesome
project!
-MC


From nicholas at montrealconsultoria.com.br  Sat Jan 31 08:28:04 2009
From: nicholas at montrealconsultoria.com.br (Nicholas Santos)
Date: Sat, 31 Jan 2009 13:28:04 -0300
Subject: [Freeswitch-dev] Freeswitch on Debian repository
Message-ID: <8171bb3b0901310828i72b9c0dj389b9ed7bc791540@mail.gmail.com>

Hi, I'm the current owner of the Freeswitch package on Debian Testing
(Lenny).

I'm just beginning to package it, if there's something that I should know
before I package it, just say it. I'm here to do my best on this that I
consider a great step.

Thanks in advance,
Nicholas Amorim.

PS: By the way, what a great piece of software! :)
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From john at skopis.com  Sat Jan 31 09:34:13 2009
From: john at skopis.com (John Skopis)
Date: Sat, 31 Jan 2009 11:34:13 -0600
Subject: [Freeswitch-dev] [RFC] mod_entity
Message-ID: <49848B95.4050400@skopis.com>

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The module is not complete. I am sending it now for
some feedback as it might be a good time (mostly completed).

The module, mod_entity, will build all xml entities the xml interface
supports, currently it only supports directory, and dialplan (partially).

The idea is that mod_entity describing xml entities, which are just
collections of elements, queries a directory (dbi) for the values it
needs to build the xml.

mod_directory accepts a char** array of fields, and returns an ordered
char ** array of values, the dn (id) of the last entry (row), and how
many entries remain.

I would like to hear what you think, anthm especially. I know it's not
finished code but I am not sure I am in the right direction.

For this module to be at all practical the xml created needs to be
cached. Lets save this for another thread though.


http://jira.freeswitch.org/browse/FSREPO-1

- -John
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From darren at aleph-com.net  Sat Jan 31 11:40:03 2009
From: darren at aleph-com.net (Darren Wiebe)
Date: Sat, 31 Jan 2009 12:40:03 -0700
Subject: [Freeswitch-dev] ANNOUNCEMENT: FreeSWITCH 1.0.3 RC1 Available
In-Reply-To: <87f2f3b90901311127o24592759q85d9e442b94a4d75@mail.gmail.com>
References: <87f2f3b90901311127o24592759q85d9e442b94a4d75@mail.gmail.com>
Message-ID: <4984A913.1060709@aleph-com.net>

Congratulations!  I was impressed again today how easy it is to install 
Freeswitch and get it routing calls,etc.  :)

Darren Wiebe
darren at aleph-com.net

Michael Collins wrote:
> The FreeSWITCH development team is happy to announce the immediate
> availability of FreeSWITCH 1.0.3 RC1. The download is available here:
> http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz
>
> More information regarding this release candidate can be found here:
> http://freeswitch.org/node/159
>
> Please help us out by downloading and running the latest FreeSWITCH
> release. Thank you all for your continued support of this awesome
> project!
> -MC
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>   



From peder at networkoblivion.com  Sat Jan 31 13:40:23 2009
From: peder at networkoblivion.com (peder at networkoblivion.com)
Date: Sat, 31 Jan 2009 15:40:23 -0600
Subject: [Freeswitch-dev] ANNOUNCEMENT: FreeSWITCH 1.0.3 RC1 Available
In-Reply-To: <87f2f3b90901311127o24592759q85d9e442b94a4d75@mail.gmail.com>
References: <87f2f3b90901311127o24592759q85d9e442b94a4d75@mail.gmail.com>
Message-ID: <4984C547.8000902@networkoblivion.com>

Is there a way to update to this specific version if one is updating via 
cvs?  Or would I have to download the tarball?

Michael Collins wrote:
> The FreeSWITCH development team is happy to announce the immediate
> availability of FreeSWITCH 1.0.3 RC1. The download is available here:
> http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz
> 
> More information regarding this release candidate can be found here:
> http://freeswitch.org/node/159
> 
> Please help us out by downloading and running the latest FreeSWITCH
> release. Thank you all for your continued support of this awesome
> project!
> -MC
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 


From krice at freeswitch.org  Sat Jan 31 13:43:22 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 31 Jan 2009 15:43:22 -0600
Subject: [Freeswitch-dev] ANNOUNCEMENT: FreeSWITCH 1.0.3 RC1 Available
In-Reply-To: <4984C547.8000902@networkoblivion.com>
Message-ID: <C5AA221A.4CFCE%krice@freeswitch.org>

Just update to trunk...


> From: "peder at networkoblivion.com" <peder at networkoblivion.com>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Sat, 31 Jan 2009 15:40:23 -0600
> To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] ANNOUNCEMENT: FreeSWITCH 1.0.3 RC1 Available
> 
> Is there a way to update to this specific version if one is updating via
> cvs?  Or would I have to download the tarball?
> 
> Michael Collins wrote:
>> The FreeSWITCH development team is happy to announce the immediate
>> availability of FreeSWITCH 1.0.3 RC1. The download is available here:
>> http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz
>> 
>> More information regarding this release candidate can be found here:
>> http://freeswitch.org/node/159
>> 
>> Please help us out by downloading and running the latest FreeSWITCH
>> release. Thank you all for your continued support of this awesome
>> project!
>> -MC
>> 
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org




From brian at freeswitch.org  Sat Jan 31 13:46:09 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 31 Jan 2009 15:46:09 -0600
Subject: [Freeswitch-dev] ANNOUNCEMENT: FreeSWITCH 1.0.3 RC1 Available
In-Reply-To: <C5AA221A.4CFCE%krice@freeswitch.org>
References: <C5AA221A.4CFCE%krice@freeswitch.org>
Message-ID: <E0DBA647-CDD9-483D-904E-76B6453D22EB@freeswitch.org>

He's right 1.0.3RC1 is trunk at this moment.  ;)

/b

On Jan 31, 2009, at 3:43 PM, Ken Rice wrote:

> Just update to trunk...
>
>
>> From: "peder at networkoblivion.com" <peder at networkoblivion.com>
>> Reply-To: "freeswitch-dev at lists.freeswitch.org"
>> <freeswitch-dev at lists.freeswitch.org>
>> Date: Sat, 31 Jan 2009 15:40:23 -0600
>> To: "freeswitch-dev at lists.freeswitch.org"
>> <freeswitch-dev at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-dev] ANNOUNCEMENT: FreeSWITCH 1.0.3 RC1  
>> Available
>>
>> Is there a way to update to this specific version if one is  
>> updating via
>> cvs?  Or would I have to download the tarball?
>>
>> Michael Collins wrote:
>>> The FreeSWITCH development team is happy to announce the immediate
>>> availability of FreeSWITCH 1.0.3 RC1. The download is available  
>>> here:
>>> http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz
>>>
>>> More information regarding this release candidate can be found here:
>>> http://freeswitch.org/node/159
>>>
>>> Please help us out by downloading and running the latest FreeSWITCH
>>> release. Thank you all for your continued support of this awesome
>>> project!
>>> -MC
>>>
>>> _______________________________________________
>>> Freeswitch-dev mailing list
>>> Freeswitch-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Freeswitch-dev mailing list
>> Freeswitch-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> dev
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



From brian at freeswitch.org  Sat Jan 31 17:12:45 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 31 Jan 2009 19:12:45 -0600
Subject: [Freeswitch-dev] Cisco 7975G and XML options for G722
Message-ID: <0C8E2714-6D96-49DB-93CD-0D377E4DAE96@freeswitch.org>

I'm digging for the option in the XML to enable G722 on the 7975G, It  
says "Enterprise Advertise G.722, Disabled", "Device Advertise G.722,  
Use System Default"  So what options to turn G722 on?  Anyone? Anyone?

Thanks,
/b