Re: [Freeswitch-users] Passing user variables to mod_voicemail

[Anthony Minessale]

That wont work.
I'm not sure if there is a way, I cant think of one off the top of my head.


On Thu, Dec 10, 2009 at 10:10 PM, Mark Campbell-Smith <
mcampbellsm...@gmail.com> wrote:

> Hi!
>
> My voip provider provides a SOAP interface to be able to send SMS's,
> so after a voicemail is left, I want to execute a 'send sms' script.
> I don't want a separate statement in the dialplan after the voicemail
> statement because I only want to send sms's when a voicemail is
> actually left.
>
> The way I was going to do this was to modify the mailer-app to point
> to a shell script and modify the mailer-app-args to include some user
> defined variables (in conf/directory/default/*.xml).
>
> value="/usr/local/freeswitch/scripts/emailvm.sh"/>
>
>
> The shell script would do the following:
>
> emailvm.sh
>
> #$1 $2 $3 = smsaccount smspassword textmessage
> tee /tmp/vmmail | /usr/sbin/sendmail -t
> exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3
> #echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log
>
> However, if I uncomment the last line, I never see the user variables
> being passed to the shell script.  The email is sucessfully sent, but
> the sms script doesnt work.  If fact, the output of log.log is (for
> example):
>
> -f 1...@192.168.1.120 email_addr...@domain.com
>
> Any ideas if it is possible to pass user variables via mod_voicemail
> in this way?
>
> Thanks
>
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-- 
Anthony Minessale II

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AIM: anthm
MSN:anthony_miness...@hotmail.com 
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[Freeswitch-users] Passing user variables to mod_voicemail

[Mark Campbell-Smith]

Hi!

My voip provider provides a SOAP interface to be able to send SMS's,
so after a voicemail is left, I want to execute a 'send sms' script.
I don't want a separate statement in the dialplan after the voicemail
statement because I only want to send sms's when a voicemail is
actually left.

The way I was going to do this was to modify the mailer-app to point
to a shell script and modify the mailer-app-args to include some user
defined variables (in conf/directory/default/*.xml).




The shell script would do the following:

emailvm.sh

#$1 $2 $3 = smsaccount smspassword textmessage
tee /tmp/vmmail | /usr/sbin/sendmail -t
exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3
#echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log

However, if I uncomment the last line, I never see the user variables
being passed to the shell script.  The email is sucessfully sent, but
the sms script doesnt work.  If fact, the output of log.log is (for
example):

-f 1...@192.168.1.120 email_addr...@domain.com

Any ideas if it is possible to pass user variables via mod_voicemail
in this way?

Thanks

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Re: [Freeswitch-users] embedded freeswitch compatable hardware

[Joseph L. Casale]

>Anyway I sent an email to Yawarra to ask them if the net5501 computer
> is
>compatible with the TDM400 cards.

It is, people have been doing this for a while w/ astlinux:
http://www.mail-archive.com/astlinux-us...@lists.sourceforge.net/msg03048.html

jlc


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Re: [Freeswitch-users] embedded freeswitch compatable hardware

[Brian May]

On Thu, Dec 10, 2009 at 09:20:39PM -0500, Michael Jerris wrote:
> As a note, we are pretty aggressive about making sure all this stuff works
> right out of svn without any patches so it should be easy to port freeswitch
> to most platforms now.

Thats good to hear.

I am guessing this means I should use a recent version. I see there is an
Ubuntu archive, wondering if that will work with Voyage Linux. If not, I should
be able to build from the source.

Anyway I sent an email to Yawarra to ask them if the net5501 computer
 is compatible with
the TDM400 cards. There is something about a kit for the dual rack mount
computer for the TDM400, which would be good if I had a rack, and somewhere to
put a rack. So presumably this means it should work for the non-rack mount
system too.
-- 
Brian May 

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Re: [Freeswitch-users] embedded freeswitch compatable hardware

[Michael Jerris]

As a note, we are pretty aggressive about making sure all this stuff works 
right out of svn without any patches so it should be easy to port freeswitch to 
most platforms now.

Mike

On Dec 10, 2009, at 8:57 PM, Brian May wrote:

> On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote:
>> Lack of OpenZAP support might be an issue, I assume that would be
>> required to connect to an onboard analogue port... I assume I could just
>> install Debian or another distribution instead though.
> 
> This is another distribution I found:
> 
> http://linux.voyage.hk/
> 
> It comes with Asterisk out of the box, although I suspect it
> wouldn't be too hard to get Freeswitch working instead.
> -- 
> Brian May 
> 
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Re: [Freeswitch-users] embedded freeswitch compatable hardware

[Brian May]

On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote:
> Lack of OpenZAP support might be an issue, I assume that would be
> required to connect to an onboard analogue port... I assume I could just
> install Debian or another distribution instead though.

This is another distribution I found:

http://linux.voyage.hk/

It comes with Asterisk out of the box, although I suspect it
wouldn't be too hard to get Freeswitch working instead.
-- 
Brian May 

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[Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda For Dec 11

[Michael Collins]

FYI,

Here's the agenda for tomorrow's conference call:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_11

Please be ready to join at 11AM CST! :) Don't forget to bring your agenda
items, questions, and a willingness to help out with our various janitor
projects.

Thanks!
-MC
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[Freeswitch-users] never mind

[Kendall Stauffer]

Sorry, I now see it wasn't loaded, so must not come with the pre compiled.
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[Freeswitch-users] windows pre compiled asr

[Kendall Stauffer]

I downloaded yesterdays latest pre compiled and seems to works great, but I get 
invalid Asr module when trying to run pizza app.
   It seemed to come pre configured with pocketsphynx,  anything I should know 
before I spend a boat load of time on it?
   Rest seems real good,. thatks!!!
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Michael Collins]

On Thu, Dec 10, 2009 at 2:16 PM, Julian Lyndon-Smith wrote:

> Ok. The journey begins.
>
> http://makingfs.blogspot.com/
>
> Don't know if you want to add this link to the website or wiki.
>
> Julian
>
> Excellent work! Thanks,
-MC

Asterisk deadlocked
Why does it suck so badly?
Use FreeSWITCH instead
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Re: [Freeswitch-users] Route Non-Call Data to Agent Through Queue

[Andrew Thompson]

On Thu, Dec 10, 2009 at 10:26:39AM -0800, Shaun Clark wrote:
> I have an application where I would like to route both calls and other
> requests through the same queue to the same agents, for example the same
> agent might take a call and then right after that take a chat. But, the chat
> server we use is separate from our phone system.
> 
> What I would like to do is basically route some text, i.e. "new chat
> chat_id_goes_here" through to the agent. Is this possible with FreeSwitch?
> The idea being the soft-phone would receive this text and we would write
> code to catch this message do the appropriate action on our CRM. Thanks!

I did this by writing my own external queueing (in erlang) and simply
parking the calls in FS and adding them to my external queue (along with
emails, voicemails, etc). With asterisk I added a fake call to the queue
with some channel variables that referenced the external data I was
really putting in the queue and I listened for the 'BRIDGE' event on the
AMI and sent the agent the external data then.

I'm not sure mod_fifo needs to be a universal queue - but maybe you
could do what you want via api_after_bridge and uuid_chat or something
crazy? You'd have to script whatever soft-phone you're using to be smart
about that though.

Andrew

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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Julian Lyndon-Smith]

Ok. The journey begins.

http://makingfs.blogspot.com/

Don't know if you want to add this link to the website or wiki.

Julian

2009/12/10 Michael Collins :
>
>
> On Thu, Dec 10, 2009 at 1:11 PM, David Knell  wrote:
>>
>> On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote:
>>
>> > Update to latest
>> > Did you type make current yet?
>> > Tony hates build skew
>>
>> Brilliant.
>>
>> Michael Collins-san
>> Shrinks all usual advice
>> Into one Haiku.
>>
>> --Dave
>
> I love the wiki
> The docs are disorganized
> I hate the wiki
> -MC
>
>
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Michael Collins]

On Thu, Dec 10, 2009 at 1:11 PM, David Knell  wrote:

>
> On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote:
>
> > Update to latest
> > Did you type make current yet?
> > Tony hates build skew
>
> Brilliant.
>
> Michael Collins-san
> Shrinks all usual advice
> Into one Haiku.
>
> --Dave
>

I love the wiki
The docs are disorganized
I hate the wiki
-MC
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[David Knell]

On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote:

> Update to latest
> Did you type make current yet?
> Tony hates build skew

Brilliant.

Michael Collins-san
Shrinks all usual advice
Into one Haiku.

--Dave


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Re: [Freeswitch-users] embedded freeswitch compatable hardware

[Tihomir Culjaga]

ok, but how much smultaneous calls did you get on an alix board using
astlinux... for istnace, this is the question?


T.


On Thu, Dec 10, 2009 at 5:12 PM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:

> On Thu, Dec 10, 2009 at 9:26 AM, Frank Carmickle 
> wrote:
> >
> > The 330 boards are a little more power hungry but you get a dual core 64
> bit processor.  As far as I'm concerned the performance increase is well
> worth the extra money.  You still well below the power consumption of any
> other 64 bit dual core machines.
> >
> > http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383
> >
> > --FC
> >
>
>   While these are low power when compared to traditional
> desktop/server systems, they're not what I would consider to be
> embedded.  The CPU requires a fan (embedded no-no) and between the
> chipset and CPU they draw several times more power than a traditional
> embedded system.  The ALIX and Soekris boards run with 12 watt power
> supplies (12v, 1 amp).  The Atom 330 alone can draw 8 watts.  This is
> still impressive for a processor of this class but it's not what I
> would consider to be embedded, yet...
>
> I think of embedded systems like this:
>
> Blackfin - Very low power, good performance (especially for DSP), very
> difficult porting (usually)
> ARM/MIPS - Very low power, decent performance depending on
> application, mild difficulty in porting
> X86 (Geode, etc) - Pretty low power, decent performance, relative ease
> in "porting" (often none)
> Everything else - You should probably call it an "appliance", not an
> embedded system
>
>  With the correct target application and design ARM and Geode systems
> can provide more than enough CPU power for many, many practical
> applications.
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
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Re: [Freeswitch-users] Monitoring IVR pressed-options in XML IVR

[Alberto Escudero]

I want to trigger CUSTOM events via ESL "as they navigate inside" of the IVR.
The XML IVRs are generated from a GUI.

The CUSTOM events need to carry

- what IVR the user is navigating
- what option has been selected
- ideally how long they stayed listening (this can be calculated)
- and when they hang the phone

/aep
-- 
Stopping junk mailers is good for the environment

> On Thu, Dec 10, 2009 at 9:07 AM, Alberto Escudero 
> wrote:
>
>>
>> Hi,
>>
>> I am currently creating IVR using the functions provided in the XML
>> dialplan
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
>>
>> Using functions like this
>> > param="$${base_dir}/1255549537_Welcome.wav"/>
>> I can play files, etc.
>>
>> I wonder what is the smartest way to monitor (as in big brother) the
>> options selected by the user:
>>
>> I assume that I can include an entry of the type:
>> 
>> and include in foo.js the code to track the selection.
>>
>> But I wonder if this is the best approach
>>
>> /aep
>>
>> Are you trying to do some sort of live monitoring as it happens (i.e.
>> while
> the call is live) or do you just want a record of the digits they pressed?
> -MC
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[Freeswitch-users] Route Non-Call Data to Agent Through Queue

[Shaun Clark]

I have an application where I would like to route both calls and other
requests through the same queue to the same agents, for example the same
agent might take a call and then right after that take a chat. But, the chat
server we use is separate from our phone system.

What I would like to do is basically route some text, i.e. "new chat
chat_id_goes_here" through to the agent. Is this possible with FreeSwitch?
The idea being the soft-phone would receive this text and we would write
code to catch this message do the appropriate action on our CRM. Thanks!
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Michael Collins]

On Thu, Dec 10, 2009 at 10:17 AM, Brian West  wrote:

> Use the BKW method... three to four word sentences to describe what to
> do... its very poetic!  Or is that haiku?
>
> /b
>

Update to latest
Did you type make current yet?
Tony hates build skew
-MC
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Re: [Freeswitch-users] Does FS support STUN by default?

[Michael Jerris]

we also support natpmp and static ip setting.

Mike

On Dec 10, 2009, at 12:21 PM, Fred-145 wrote:

> 
> Thanks for the clarification. So it's either UPnP or STUN/port-mapping.


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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Brian West]

Use the BKW method... three to four word sentences to describe what to  
do... its very poetic!  Or is that haiku?

/b

On Dec 10, 2009, at 11:53 AM, Michael Collins wrote:

>
> I was just thinking of some way to learn FS gradually and  
> effectively. The
> frequent problem with wiki's, is that the quality of articles is  
> uneven and
> they don't have a good layout. But then, writing documentation is  
> hard and
> time-consuming :-/
>
> Amen, brothah! :)
> -MC


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Re: [Freeswitch-users] Monitoring IVR pressed-options in XML IVR

[Michael Collins]

On Thu, Dec 10, 2009 at 9:07 AM, Alberto Escudero  wrote:

>
> Hi,
>
> I am currently creating IVR using the functions provided in the XML
> dialplan
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
>
> Using functions like this
>  param="$${base_dir}/1255549537_Welcome.wav"/>
> I can play files, etc.
>
> I wonder what is the smartest way to monitor (as in big brother) the
> options selected by the user:
>
> I assume that I can include an entry of the type:
> 
> and include in foo.js the code to track the selection.
>
> But I wonder if this is the best approach
>
> /aep
>
> Are you trying to do some sort of live monitoring as it happens (i.e. while
the call is live) or do you just want a record of the digits they pressed?
-MC
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Re: [Freeswitch-users] Routing calls to Another FS server

[Michael Collins]

On Thu, Dec 10, 2009 at 8:42 AM, Otis  wrote:

> I have 2 FS servers FS1 (aka medion) and FS3 (callweaver). These are set
> as gateways and register with each other. I wanted all users on FS1 to
> dial those on FS3 with  prefix 33  ie extn 1001 on FS3 will be dialed as
> 331001 on FS1.
>
> I have a dialplan as follows in .../dialplan/default/ callweaver.xml
>
> > 
> > 
> > 
> > 
> > 
>
> I have also used
>
> 
>  
>
>  
> 
>
> I have also used the line  data="sofia/gateway/outbound.callweaver/$1"/>
> in place of  data="sofia/profilename/$...@192.168.1.110"/>
> without any joy.
>
> I am getting error  Not - found from the client.  I am registered as
> 1001 on FS1.
>

Otis,

I would recommend that you turn on debugging and capture the console output.
Put it in a pastebin and then respond to this thread with a link to the
pastebin URL. FYI, here's a handy page that will give you some nice pointers
on how to gather information when you're doing troubleshooting:

http://wiki.freeswitch.org/wiki/Reporting_Bugs

It has detailed instructions on getting console output, dumping to pastebin,
etc. If you want to get really fancy you can also try Tony's handy-dandy
debug Perl script: libs/esl/perl/logger.pl under the FS source directory.
Run the script, make your call, then hit ctrl-C. The script will upload the
log to pastebin for you and tell you the URL. :)

-MC
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Michael Collins]

On Thu, Dec 10, 2009 at 3:40 AM, Fred-145  wrote:

>
> No publisher, although uploading and selling books (deadtree or online) is
> easy with companies like www.lulu.com
>
> I was just thinking of some way to learn FS gradually and effectively. The
> frequent problem with wiki's, is that the quality of articles is uneven and
> they don't have a good layout. But then, writing documentation is hard and
> time-consuming :-/
>

Amen, brothah! :)
-MC
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Michael Collins]

On Thu, Dec 10, 2009 at 2:13 AM, Julian Lyndon-Smith wrote:

> Sometime next week I hopefully am going to start a document that
> follows my progress in setting up a FS system from scratch, with all
> the pitfalls and successes. A kinds of "warts and all" story.
> Alongside this "blog" (for want of a better word) I will also then
> document the steps needed to get it working (a howto guide,
> effectively).
>
> I am a long time * user (2004), so my mindset is kind of skewed - but
> perhaps that would be beneficial for other * users looking at
> implementing FS.
>
> Most of our config and dialplan is generated by using res_config_curl,
> and we use things like call listening, conferencing, parking and
> queues. We do use queues in a slightly odd manner (we add 1 agent, and
> call a local channel). When this channels is called, we use curl to
> get our application to return the most appropriate agent to actually
> call).
>
> We also use * as a power dialler, making upwards of 400,000 call
> attempts per month. Not massive, but not tiny either.
>
> Hopefully, this will be of use to both FS and * users. What would be
> great is that if other people follow my progress, and make suggestions
> as and when I hit a brick wall :)
>
> What would be best for this ? A blog ? Or a wiki page ?
>

Julian,

First off, welcome to the FreeSWITCH fold! Thank you for your willingness
not only to try things but to document them as well. I like the idea of a
blog to tell the story. After you're done then you might consider a wiki
page that is more of "just the facts, ma'am" on how you translated Asterisk
configs to a FreeSWITCH setup. Feel free to ask myself or anyone here or on
IRC for assistance.

-MC (IRC: mercutioviz)
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Re: [Freeswitch-users] Does FS support STUN by default?

[Fred-145]

Thanks for the clarification. So it's either UPnP or STUN/port-mapping.
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[Freeswitch-users] Monitoring IVR pressed-options in XML IVR

[Alberto Escudero]

Hi,

I am currently creating IVR using the functions provided in the XML dialplan
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr

Using functions like this

I can play files, etc.

I wonder what is the smartest way to monitor (as in big brother) the
options selected by the user:

I assume that I can include an entry of the type:

and include in foo.js the code to track the selection.

But I wonder if this is the best approach

/aep

-- 
Stopping junk mailers is good for the environment




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Re: [Freeswitch-users] Routing calls to Another FS server

[Otis]

Otis wrote:
I have 2 
FS servers FS1 (aka medion) and FS3 (callweaver). These are set as 
gateways and register with each other. I wanted all users on FS1 to 
dial those on FS3 with  prefix 33  ie extn 1001 on FS3 will be dialed 
as 331001 on FS1.


I have a dialplan as follows in .../dialplan/default/ callweaver.xml









I have also used


 
   data="sofia/profilename/$...@192.168.1.110"/>

 


I have also used the line data="sofia/gateway/outbound.callweaver/$1"/>
in place of data="sofia/profilename/$...@192.168.1.110"/>

without any joy.

I am getting error  Not - found from the client.  I am registered as 
1001 on FS1.


Please how do I make all users use this dial plan  and may I know 
which version of all those stated above is right.  All are in the 
...dialplan/default directory.  called callweaver.xml


Should it have a particular name either than the gateway name ?

Thanks for your time once again





Sorry I forgot to add that where it says profilename I have *callweaver* 
which is the profile name of the gateway in  
/conf/sip_profiles/external/callweaver.xml


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Re: [Freeswitch-users] Invite local number into a conference - codec problem

[Anthony Minessale]

set absolute_codec_string to whatever codec you want to offer in the {} on
the bridge string


On Thu, Dec 10, 2009 at 8:26 AM, Peter P GMX  wrote:

> Hello,
>
> I try to invite a user into a conference by
> loopback/255 8000 Conference
> 255 is the user, I invite the user via loopback as that way I can also
> invite external numbers.
>
> It processes the user's local dialplan correctly (as if the user was
> normally dialled), however it only offers L16 codec, so the Phone fails.
> I can see no codec negociation on the debug console.
> If I call the phone from another phone, then codec negociation is taking
> place.
> If I invite an external PSTN user into the conference then codecs are
> set correctly (L16+PCMA+PCMU etc)
>
> Is there a way to explicitely set the codec for the conference?
>
>  is not set is,
> still commented in the internal profile.
> In vars.conf.xml only only PCMA and PCMU are set.
>
> Best regards
> Peter
>
>
>
>
>
>
> ___
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>



-- 
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
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[Freeswitch-users] Routing calls to Another FS server

[Otis]

I have 2 FS servers FS1 (aka medion) and FS3 (callweaver). These are set 
as gateways and register with each other. I wanted all users on FS1 to 
dial those on FS3 with  prefix 33  ie extn 1001 on FS3 will be dialed as 
331001 on FS1.

I have a dialplan as follows in .../dialplan/default/ callweaver.xml

> 
> 
> 
> 
> 
 
I have also used


  

  


I have also used the line 
in place of 
without any joy.

I am getting error  Not - found from the client.  I am registered as 
1001 on FS1.

Please how do I make all users use this dial plan  and may I know which 
version of all those stated above is right.  All are in the 
...dialplan/default directory.  called callweaver.xml

Should it have a particular name either than the gateway name ?

Thanks for your time once again


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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Anthony Minessale]

Don't worry.
I was an asterisk developer/volunteer in 2003.  I still managed to figure it
out. ;)


On Thu, Dec 10, 2009 at 4:13 AM, Julian Lyndon-Smith wrote:

> Sometime next week I hopefully am going to start a document that
> follows my progress in setting up a FS system from scratch, with all
> the pitfalls and successes. A kinds of "warts and all" story.
> Alongside this "blog" (for want of a better word) I will also then
> document the steps needed to get it working (a howto guide,
> effectively).
>
> I am a long time * user (2004), so my mindset is kind of skewed - but
> perhaps that would be beneficial for other * users looking at
> implementing FS.
>
> Most of our config and dialplan is generated by using res_config_curl,
> and we use things like call listening, conferencing, parking and
> queues. We do use queues in a slightly odd manner (we add 1 agent, and
> call a local channel). When this channels is called, we use curl to
> get our application to return the most appropriate agent to actually
> call).
>
> We also use * as a power dialler, making upwards of 400,000 call
> attempts per month. Not massive, but not tiny either.
>
> Hopefully, this will be of use to both FS and * users. What would be
> great is that if other people follow my progress, and make suggestions
> as and when I hit a brick wall :)
>
> What would be best for this ? A blog ? Or a wiki page ?
>
> Julian
>
> 2009/12/9 Brian West :
> > That is what is nice about our community I'm more than willing to answer
> the
> > questions if you document them... as are many others in the core
> team...we
> > just have a lot to do and I think the best repayment is documentation! ;)
> > /b
> > On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote:
> >
> > On Thu, Dec 10, 2009 at 11:07 AM, Brian West 
> wrote:
> >
> > Visit the friday meetings and we can help if you document it.  ;)
> >
> >
> > I would be willing to lend a hand with the documentation but I know so
> > little (a complete freeswitch noob). For example I was trying to
> > figure out how to tell if an extension was set up "show dialplan in
> > asterisk".  I could not find this anywhere. If I find out I would be
> > happy to add it to the rosetta stone.
> >
> > I am currently working on getting outbound socket working. Once I get
> > it going I would be happy to add it to the relevant section of the
> > wiki (in this case ruby).
> >
> >
> > ___
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> >
> >
>
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-- 
Anthony Minessale II

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ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_miness...@hotmail.com 
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Re: [Freeswitch-users] embedded freeswitch compatable hardware

[Kristian Kielhofner]

On Thu, Dec 10, 2009 at 9:26 AM, Frank Carmickle  wrote:
>
> The 330 boards are a little more power hungry but you get a dual core 64 bit 
> processor.  As far as I'm concerned the performance increase is well worth 
> the extra money.  You still well below the power consumption of any other 64 
> bit dual core machines.
>
> http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383
>
> --FC
>

  While these are low power when compared to traditional
desktop/server systems, they're not what I would consider to be
embedded.  The CPU requires a fan (embedded no-no) and between the
chipset and CPU they draw several times more power than a traditional
embedded system.  The ALIX and Soekris boards run with 12 watt power
supplies (12v, 1 amp).  The Atom 330 alone can draw 8 watts.  This is
still impressive for a processor of this class but it's not what I
would consider to be embedded, yet...

I think of embedded systems like this:

Blackfin - Very low power, good performance (especially for DSP), very
difficult porting (usually)
ARM/MIPS - Very low power, decent performance depending on
application, mild difficulty in porting
X86 (Geode, etc) - Pretty low power, decent performance, relative ease
in "porting" (often none)
Everything else - You should probably call it an "appliance", not an
embedded system

  With the correct target application and design ARM and Geode systems
can provide more than enough CPU power for many, many practical
applications.

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

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Re: [Freeswitch-users] Does FS support STUN by default?

[Rupa Schomaker]

STUN is not a way to open ports in a manner in which sheilds up would
detect.

http://en.wikipedia.org/wiki/STUN

UPNP is what you want if you want to open
ports.

STUN is just a method for figuring out how to do nat traversal.  STUN
"method" is initiated by the process on the inside of the firewall at
connection establish time.  It will do no good for the listen case which is
what you are checking for.

If you want to use stun, then you need to forward the listen ports manually
(5060, 5080 - UDP and TCP).

On Thu, Dec 10, 2009 at 8:04 AM, Fred-145  wrote:

>
> Hello
>
> I wanted to check if my ADSL modem worked with STUN, so I left its "UPNP
> activity" option unchecked, ran FreeSwitch, and used eg. Shields Up
> (www.grc.com) to check if UDP5080 (and possibly UDP5060) were opened...
> which SU says no.
>
> Does it mean that...
> - by default, FS doesn't use STUN
> - or my modem doesn't support STUN, and I must either enable UPnP or map
> ports (UDP5080 and some UDP ports for RTP/RTPC) statically?
>
> Thank you.
> --
> View this message in context:
> http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26727762.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] embedded freeswitch compatable hardware

[Frank Carmickle]

On Thu, Dec 10, Fred-145 wrote:
> 
> 
> Frank Carmickle wrote:
> > A board with an atom 330 on it would probably do the trick for you.  There
> > are a few made by Intel and Supermicro that look pretty nice.  There were
> > some other people on the list looking to use them.  Maybe we can get a
> > report from someone.
> 
> Intel came up with the D945GSEJT, which is totally fanless and has an
> embedded DC/DC, so all you have to add is an external AC/DC power brick,
> some RAM, a PCI riser to save space, and either a hard-disk or a
> CompactFlash + IDE adaptor. I'm thinking of building one with a
> Digium-compatible PCI card.
> 
> www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm

The 330 boards are a little more power hungry but you get a dual core 64 bit 
processor.  As far as I'm concerned the performance increase is well worth the 
extra money.  You still well below the power consumption of any other 64 bit 
dual core machines.  

http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383

--FC

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[Freeswitch-users] Invite local number into a conference - codec problem

[Peter P GMX]

Hello,

I try to invite a user into a conference by
loopback/255 8000 Conference
255 is the user, I invite the user via loopback as that way I can also
invite external numbers.

It processes the user's local dialplan correctly (as if the user was
normally dialled), however it only offers L16 codec, so the Phone fails.
I can see no codec negociation on the debug console.
If I call the phone from another phone, then codec negociation is taking
place.
If I invite an external PSTN user into the conference then codecs are
set correctly (L16+PCMA+PCMU etc)

Is there a way to explicitely set the codec for the conference?

 is not set is,
still commented in the internal profile.
In vars.conf.xml only only PCMA and PCMU are set.

Best regards
Peter






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Re: [Freeswitch-users] [vars.xml] default_password=1234?

[Fred-145]

Ah, makes sense:

conf/directory/default/1000.xml:



  

Thanks for the tip.
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[Freeswitch-users] Does FS support STUN by default?

[Fred-145]

Hello

I wanted to check if my ADSL modem worked with STUN, so I left its "UPNP
activity" option unchecked, ran FreeSwitch, and used eg. Shields Up
(www.grc.com) to check if UDP5080 (and possibly UDP5060) were opened...
which SU says no.

Does it mean that...
- by default, FS doesn't use STUN
- or my modem doesn't support STUN, and I must either enable UPnP or map
ports (UDP5080 and some UDP ports for RTP/RTPC) statically?

Thank you.
-- 
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Re: [Freeswitch-users] [vars.xml] default_password=1234?

[Brian West]

please look in conf/directory/default/*.xml

/b

On Dec 10, 2009, at 7:40 AM, Fred-145 wrote:

>
> Hello
>
> I'm going through the various XML files, and noticed this first line  
> in
> vars.xml.
>
> 
>
> What is this password used for?
>
> Thank you


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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS

[Brian West]

I have confirmed it works with Polycom, Snom and a few others   
polycom is the hardest to set due to having to put the ca cert into  
the phone... but other than that its good.

/b

On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote:

> An intermediate report:
>
> Audiocodes: TLS works only on outgoing requests, incoming ones are  
> ignored. I am waiting for Audiocodes' help in order to debug it.
> SRTP: worked when no TLS is active. When TLS is active the call is  
> disconnected when the remote party answers. Still debugging it.
>
> VegaStream Europa-50: SRTP works. Waiting for Vega for instructions  
> how to enable TLS from the WEB interface.
>
>  Regards, __Yehavi:


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[Freeswitch-users] [vars.xml] default_password=1234?

[Fred-145]

Hello

I'm going through the various XML files, and noticed this first line in
vars.xml.



What is this password used for?

Thank you
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Re: [Freeswitch-users] embedded freeswitch compatable hardware

[Fred-145]

Mark Campbell-Smith wrote:
> I use a dectop by Data Evolution... Its cheap at ~$100.  I have it
> running debian lenny and FS... works well for me.

Thanks for the tip, although this type of box doesn't have a PCI slot, so
the only way to connect FS to the PSTN is through a VoIP provider (or a
Linksys 3102, with the usual, possible echo issues).

In the same vein as the decTop : http://en.wikipedia.org/wiki/SheevaPlug

Since FS can be compiled to ARM, Freeswitch might be able to run on this
device:
http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux
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Re: [Freeswitch-users] embedded freeswitch compatable hardware

[Mark Campbell-Smith]

I use a dectop by Data Evolution... Its cheap at ~$100.  I have it
running debian lenny and FS... works well for me.

http://www.dataevolution.com/dectop%20info%202.htm

http://www.gadgettastic.com/2007/08/18/dectop-the-100-pc/


On Thu, Dec 10, 2009 at 10:45 PM, Fred-145  wrote:
>
>
> Frank Carmickle wrote:
>> A board with an atom 330 on it would probably do the trick for you.  There
>> are a few made by Intel and Supermicro that look pretty nice.  There were
>> some other people on the list looking to use them.  Maybe we can get a
>> report from someone.
>
> Intel came up with the D945GSEJT, which is totally fanless and has an
> embedded DC/DC, so all you have to add is an external AC/DC power brick,
> some RAM, a PCI riser to save space, and either a hard-disk or a
> CompactFlash + IDE adaptor. I'm thinking of building one with a
> Digium-compatible PCI card.
>
> www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm
> --
> View this message in context: 
> http://old.nabble.com/embedded-freeswitch-compatable-hardware-tp26721589p26725918.html
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>
>
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Re: [Freeswitch-users] embedded freeswitch compatable hardware

[Fred-145]

Frank Carmickle wrote:
> A board with an atom 330 on it would probably do the trick for you.  There
> are a few made by Intel and Supermicro that look pretty nice.  There were
> some other people on the list looking to use them.  Maybe we can get a
> report from someone.

Intel came up with the D945GSEJT, which is totally fanless and has an
embedded DC/DC, so all you have to add is an external AC/DC power brick,
some RAM, a PCI riser to save space, and either a hard-disk or a
CompactFlash + IDE adaptor. I'm thinking of building one with a
Digium-compatible PCI card.

www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm
-- 
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Fred-145]

No publisher, although uploading and selling books (deadtree or online) is
easy with companies like www.lulu.com

I was just thinking of some way to learn FS gradually and effectively. The
frequent problem with wiki's, is that the quality of articles is uneven and
they don't have a good layout. But then, writing documentation is hard and
time-consuming :-/
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

[Julian Lyndon-Smith]

Sometime next week I hopefully am going to start a document that
follows my progress in setting up a FS system from scratch, with all
the pitfalls and successes. A kinds of "warts and all" story.
Alongside this "blog" (for want of a better word) I will also then
document the steps needed to get it working (a howto guide,
effectively).

I am a long time * user (2004), so my mindset is kind of skewed - but
perhaps that would be beneficial for other * users looking at
implementing FS.

Most of our config and dialplan is generated by using res_config_curl,
and we use things like call listening, conferencing, parking and
queues. We do use queues in a slightly odd manner (we add 1 agent, and
call a local channel). When this channels is called, we use curl to
get our application to return the most appropriate agent to actually
call).

We also use * as a power dialler, making upwards of 400,000 call
attempts per month. Not massive, but not tiny either.

Hopefully, this will be of use to both FS and * users. What would be
great is that if other people follow my progress, and make suggestions
as and when I hit a brick wall :)

What would be best for this ? A blog ? Or a wiki page ?

Julian

2009/12/9 Brian West :
> That is what is nice about our community I'm more than willing to answer the
> questions if you document them... as are many others in the core team...we
> just have a lot to do and I think the best repayment is documentation! ;)
> /b
> On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote:
>
> On Thu, Dec 10, 2009 at 11:07 AM, Brian West  wrote:
>
> Visit the friday meetings and we can help if you document it.  ;)
>
>
> I would be willing to lend a hand with the documentation but I know so
> little (a complete freeswitch noob). For example I was trying to
> figure out how to tell if an extension was set up "show dialplan in
> asterisk".  I could not find this anywhere. If I find out I would be
> happy to add it to the rosetta stone.
>
> I am currently working on getting outbound socket working. Once I get
> it going I would be happy to add it to the relevant section of the
> wiki (in this case ruby).
>
>
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Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS

[Yehavi Bourvine]

An intermediate report:

*Audiocodes*: TLS works only on outgoing requests, incoming ones are
ignored. I am waiting for Audiocodes' help in order to debug it.
SRTP: worked when no TLS is active. When TLS is active the call is
disconnected when the remote party answers. Still debugging it.

*VegaStream Europa-50*: SRTP works. Waiting for Vega for instructions how to
enable TLS from the WEB interface.

 Regards, __Yehavi:

2009/12/4 Yehavi Bourvine 

> I'll report when I am done.
>
> So far I've enabled only SRTP and both support it.
>
>  __Yehavi:
>
> 2009/12/4 Mark Campbell-Smith 
>
>> Thanks Yehavi,
>>
>> I would be very interested to find out how your test goes... can you
>> report back after you have tested it?
>>
>> Thanks!
>>
>> On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine
>>  wrote:
>> > Hello,
>> >
>> >   I have AudioCodes MP and Vega ATA adapters. They both support SRTP;
>> they
>> > should support TLS also (will try it next week; up to now I preffered to
>> not
>> > use TLS so I can sniff the traffic and debug things).
>> >
>> >  Regards, __Yehavi:
>> >
>> > 2009/12/4 Mark Campbell-Smith 
>> >>
>> >> Cheers Gabriel.. thanks for the information.
>> >>
>> >> I'll look at the Mediatrix ATA's as an alternative - has anyone had
>> >> experience with those and TLS/SRTP?
>> >>
>> >>
>> >> On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri  wrote:
>> >> > The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are
>> the
>> >> > Grandstream and Mediatrix devices (although I've never tried either
>> >> > one with FreeSWITCH).
>> >> >
>> >> > I've personally never had any good experience with the Grandstream
>> >> > ATAs. The Mediatrix ATAs are OK devices, but I've never personally
>> >> > tested them with SRTP w/SDES and FreeSWITCH, but supposedly they
>> >> > support it (so says their marketing material and docs).
>> >> >
>> >> > I'd see if Cisco has any plans to add support for it to the ATAs.
>> Next
>> >> > time I see our Cisco SE, I'll try to poke him about it.
>> >> >
>> >> > Gabe
>> >> >
>> >> > On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith
>> >> >  wrote:
>> >> >> Quote: Cisco/Linksys SPA series ATAs do not support SDES key
>> exchange
>> >> >> to appropriately support SRTP and FreeSWITCH
>> >> >>
>> >> >> I'll check with Cisco regarding their implementation then and try to
>> >> >> find out when/if they will support standard SRTP encryption.
>> >> >>
>> >> >>
>> >> >> So, back to my origianal question then.  Are there any ATA's that
>> >> >> support TLS AND SRTP with FreeSwitch?
>> >> >>
>> >> >>
>> >> >> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri 
>> wrote:
>> >> >>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key
>> >> >>> exchange to appropriately support SRTP and FreeSWITCH. They do
>> their
>> >> >>> proprietary Sipura key exchange only, not sure if Cisco plans on
>> >> >>> upgrading the firmware to ever support SDES on the ATAs. They added
>> >> >>> support for SDES to their IP Phones about 1 year ago, but nothing
>> has
>> >> >>> happened with the ATAs as of yet.
>> >> >>>
>> >> >>> Gabe
>> >> >>>
>> >> >>>
>> >> >>> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith
>> >> >>>  wrote:
>> >>  Hi All,
>> >> 
>> >>  I managed to borrow a SPA3102 with the latest firmware and have
>> got
>> >>  it
>> >>  to register using TLS, but I am still struggling with SRTP.  Has
>> >>  anyone managed to get SRTP working with the Linksys devices and if
>> >>  so,
>> >>  can they direct me on how to do this.
>> >> 
>> >>  I have generated a mini-certificates and SRTP Private Key using
>> the
>> >>  gen-mc tool found at
>> >> 
>> >> 
>> http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3
>> .
>> >>   However, when ever I initiate a call from the SPA, I can see that
>> >>  the
>> >>  call is not encrypted.
>> >> 
>> >>  Help appreciated.
>> >> 
>> >>  Thanks!
>> >> 
>> >> 
>> >>  On Sat, Nov 28, 2009 at 6:31 AM, eman  wrote:
>> >> > Check out the Linksys SPA2102
>> >> >
>> >> > On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith
>> >> >  wrote:
>> >> >>
>> >> >> The only ATA mentioned on the WIKI that supports TLS/SRTP is the
>> >> >> Grandstream HandyTone 503.  But, again according to the wiki,
>> that
>> >> >> doesn't seem to behave to well with TLS ...
>> >> >>
>> >> >> On Wed, Nov 25, 2009 at 7:14 PM, Jason White <
>> ja...@jasonjgw.net>
>> >> >> wrote:
>> >> >> > Mark Campbell-Smith  wrote:
>> >> >> >> Does the SPA3102 support TLS or only SRTP?
>> >> >> >
>> >> >> > I don't know, but supporting only SRTP would be ridiculous,
>> since
>> >> >> > the
>> >> >> > keys
>> >> >> > would then be transmitted in the clear and therefore amenable
>> to
>> >> >> > interception.
>> >> >> > SRTP requires the SIP channel to be e

Re: [Freeswitch-users] embedded freeswitch compatable hardware

[Tihomir Culjaga]

Kristian,

from your experience, supposed we go for net5501 + a 4 - 8 FXS card, what is
the maximum simultaneous calls that this box can handle of course using g729
codec?

I used blackgin (IP08), alix2d3... and all of them were giving up on 6-7
simultaneous calls.
To be honest, i didnt run AstLinux on alix i used voyage instead but
anyhow... this seems to be the limit.


what i'm looking for it an appliance to run 2-16 FXS on it any
suggestion?

T.


On Thu, Dec 10, 2009 at 4:47 AM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:

> Brian,
>
>  I have been making efforts to fully support FreeSWITCH in AstLinux.
> Our primary targets are low powered x86 boards like the Soekris and
> Alix.  x86, powerful enough, cheap enough (as low as $100), and about
> 12 watts.  Not bad.
>
>  The Soekris net5501 and standard case will (I believe) take a full
> height card.  Then again you could use any board and get an external
> SIP gateway (ATA).  We don't currently support OpenZAP with FS in
> AstLinux but I'd love to add support for it eventually.
>
>  I'm currently working with the FS devs on getting some issues in
> trunk resolved to get cross compiling working again.  Until then you
> can find ISOs with FreeSWITCH and AstLInux here if you'd like to check
> it out:
>
> http://mirror.astlinux.org/freeswitch/daily/
>
>  Let me know what you think.
>
> On Wed, Dec 9, 2009 at 7:55 PM, Brian May
>  wrote:
> > Hello,
> >
> > I asked this question on my local linux user group mailing list, and got
> the
> > recommendation to ask here.
> >
> > Anyway, at the moment I am running Asterisk on an IP04 embedded system.
> > http://www.rowetel.com/ucasterisk/ip04.html
> >
> > It works well most of the time, however there are some bugs that do,
> under
> > circumstances lead to less then desirable behaviour (such as on some
> occasions
> > which I don't fully understand sometimes the remote system fails to
> generate
> > any audio packets when there is no audio - almost like silence
> suppression was
> > supported by the remote system - and asterisk fails to generate any audio
> > packets in return; on another slower computer running the same SIP
> software and
> > on the same network everything works fine; as far as I can tell the
> software -
> > twinkle - doesn't even support silence suppression).
> >
> > I suspect at least some - if not all - of the issues I have encountered
> may be
> > resolved with Freeswitch, however I don't really want to replace my
> small,
> > energy efficient, embedded system, with a large, power hungry computer
> system.
> > Overkill.
> >
> > An added complication is I need at least 1 analogue port to connect to
> the
> > Australian based telephone line (2 ports exchange ports and 1 extension
> port
> > would be ideal but not essiential).
> >
> > Unfortunately, I have been told that the IP04 hardware isn't compatable
> with
> > the requirements of Freeswitch. Such as not having a MMU. So there
> doesn't
> > appear to be much effort porting Freeswitch to IP04 as a result.
> >
> > I do have a spare TDM400p card, although as it is full height, suspect
> this
> > isn't going to help.
> >
> > Are there any other good alternatives?
> >
> > Thanks.
> > --
> > Brian May 
> >
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>
>
>
> --
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> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
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