Re: [Freeswitch-users] A little Q931 tool

[Helmut Kuper]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

ups, main message of me wasn't in my mail. Of course you can have it if
someone wants it.

regards
Helmut


On 24.09.2009 19:22, Michael Collins wrote:
> Excellent, thanks!
> -MC
> 
> On Thu, Sep 24, 2009 at 4:50 AM, Helmut Kuper  > wrote:
> 
> Hello,
> 
> I ported my perl based FS-Logfile-Q931-HexDumps-to-pcap script to C
> (linux). It reads FS's logfile (loglevel DEBUG) grabs the Q931 hex dumps
> and puts them in a .pcap file which is directly readable and decodeable
> by Wireshark/tshark.
> 
> The big plus here is, that you are able to get an isdn Q931 trace even
> of a call in the past (as long as you have FS in DEBUG loglevel).
> 
> 
> regards
> Helmut

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Re: [Freeswitch-users] [Freeswitch-dev] Status of ubuntu/debian packages.

[Jason White]

[Just catching up on this thread.]
William King  wrote:
> I would be more than happy to share the code I use.
> 
> Here is the git repo:
> 
> http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/

When you would like your changes to the Debian build infrastructure in
FreeSWITCH to be tested on Debian Sid, I'll gladly volunteer, excluding any
modules that depend on proprietary software that I don't have and don't want,
e.g., Skype.

I would also like to see these changes integrated into the FreeSWITCH
repository to replace what is currently in the debian directory, once you have
a version that is well tested.


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Re: [Freeswitch-users] How to play more than one voice file in play_and_get_digits function

[Brian West]

Either work... file string is new... we use phrase macros in the  
default IVR menu... its a nice way to make something internationalized.

/b

On Sep 24, 2009, at 11:56 PM, velusamy velu wrote:

> Dear All,
>   mod_file_string is working for me. Is there any problem by  
> using mod_file-string? If it so, how can use phrase macros in perl  
> using ESL::IVR module?
>
> Please provide your valuable idea...
>
>
>


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Re: [Freeswitch-users] How do I get freeswitch variable which has DTMF values.

[Brian West]

Or use the socket without async so that it blocks till the action is  
complete.

/b

On Sep 25, 2009, at 12:13 AM, velusamy velu wrote:

> To get the freeswitch variable I used getVar subroutine which is  
> defined in ESL::IVR.pm file. When I print that digits, Perl program  
> prints empty value while playing the menu itself. If I need to get  
> the DTMF value I need to wait the perl program.


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[Freeswitch-users] How do I get freeswitch variable which has DTMF values.

[velusamy velu]

Dear All,
 I am doing IVR by using perl ESL libraries. I have used ESL::IVR
module. I get the DTMF by using playAndGetDigits subroutine which is defined
in ESL::IVR.pm. The DTMF digit stored in freeswitch "digit" variable.

To get the freeswitch variable I used getVar subroutine which is defined
in ESL::IVR.pm file. When I print that digits, Perl program prints empty
value while playing the menu itself. If I need to get the DTMF value I need
to wait the perl program.

My question is why the Perl program prints the empty value before
executing the getVar function?
How can get the DTMF value by using getVar function without waiting in
the Perl program?

Please any one help me in this problem

Thanks,
Velusamy.
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Re: [Freeswitch-users] How to play more than one voice file in play_and_get_digits function

[velusamy velu]

Dear All,
  mod_file_string is working for me. Is there any problem by using
mod_file-string? If it so, how can use phrase macros in perl using ESL::IVR
module?

Please provide your valuable idea...





On Thu, Sep 24, 2009 at 10:25 PM, Diego Viola  wrote:

> Use phrase macros as Brian said.
>
> On Thu, Sep 24, 2009 at 1:09 PM, Brian West  wrote:
>
>> You can also use phrase macros.  (and no its not just for TTS ;)  )
>> /b
>>
>> On Sep 24, 2009, at 2:02 AM, Michael Collins wrote:
>>
>> Make sure that mod_file_string is built and loaded and then try the syntax
>> that is described here:
>> http://wiki.freeswitch.org/wiki/Mod_file_string#Examples
>>
>> Instead of a comma separated list you can use ! and be sure NOT to put a
>> space after the ! because the function delimits the arguments with spaces.
>> Try something like this:
>>
>> $conn->execute("play_and_get_digits", 1 1 1000 #
>> /usr/local/freeswitch/en/us/callie/sounds/ivr/please.wav!/usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav
>> /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+);
>>
>> Let us know how it goes.
>> -MC
>>
>>
>>
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Re: [Freeswitch-users] Status of ubuntu/debian packages.

[William King]

It has been removed from the dependencies. Thanks go to the reporter for
finding the extra depends.

A new round of builds just went out and built. Let me know if you find
something else. Also mod_skypiax should be available.

-William King

Michael Jerris wrote:
> I can confirm you should not need the swig dependency at all for  
> anything.
>
> Mike
>
> On Sep 24, 2009, at 1:49 PM, William King wrote:
>
>   
>> Hmm... That is interesting... swig is needed I believe only for the
>> mod_perl or the esl modules. I'll find out more information and put it
>> on the correct package.
>>
>> I will also update the mod_skypiax config files in the *.install  
>> files.
>>
>> -William King
>>
>> Dmitry Bely wrote:
>> 
>>> On Thu, Sep 24, 2009 at 2:14 AM, William King  
>>>  wrote:
>>>
>>>   
 Sure, post it here and I'll add it in the next build in a few hours.

 
>>> See attached file.
>>>
>>> Unfortunately mod_skypiax author did not placed config files
>>> (skypiax.conf.xml, skypiax.X.conf) into conf/autoload_configs, so  
>>> they
>>> are not included into freeswitch-config and should be added manually.
>>>
>>> BTW, why swig is a dependency for the source package? I recall  
>>> Brian's
>>> post where he insists that swig is never needed to build Freeswitch.
>>>
>>>
>>>   
 -William King

 Dmitry Bely wrote:

 
> On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle    
>> wrote:
>> 
>   
>> On Wed, Sep 23, Dmitry Bely wrote:
>>
>>
>> 
>>> Can you enable mod_skypiax in your debian package?
>>>
>>>
>>>   
>> We will be enabling as much as we can cleanly build on debian/ 
>> ubuntu.  There will be a lot more to come.  We will be breaking  
>> the mods and end points in to different packages so that you can  
>> install what you like.  If you have something you would like to  
>> see in the package let us know.  Also patches are welcome.
>>
>>
>> 
> Well, mod_skypiax just requires trivial one-line addition to
> debian/rules and debian/freeswitch.install. It builds OK. If the  
> patch
> is required I can post it here.
>
>   
>>> - Dmitry Bely
>>>
>>> 
>>>
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Re: [Freeswitch-users] instant messaging

[Michael Jerris]

I think you need to enable presence as well as have the right profile  
aliases in place (they are in the default configuration).

Mike

On Sep 24, 2009, at 4:20 PM, RobertT wrote:

> Hi guys!
>
> I'm considering to use SIMPLE protocol for IM in my application, but  
> get a following error trying to send a message from one registered  
> user to another:
> [ERR] sofia_presence.c:93 Chat proto [sip]
> from [1...@xx.xxx.xx.xx]
> to [1...@xx.xxx.xx.xx]
> 11
> Invalid Profile xx.xxx.xx.xx
>
> Should presence be enabled in order SIMPLE to work? What additional  
> steps do I have to complete in order to make presence work in FS  
> besides setting "manage_presence" param in SIP profile to true?
>
> Best regards, Robert.
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Re: [Freeswitch-users] mod_cdr_csv missing timestamps in A-LEG

[Michael Jerris]

Can you get these same values in xml-cdr?  I don't think csv was ever  
intended to work with different cdrs for a and b leg, it was more  
intended as a more familiar interface for those coming over from  
asterisk.


Mike

On Sep 24, 2009, at 6:10 AM, Tihomir Culjaga wrote:


hello,

i'm on latest trunk and for some reason i cannot get timestamps  
dumped in my cdrs. I use mod_cdr_csv with default settings plus i  
enabled to get both a and b legs dumped.



cdr_csv.conf.xml:


  








  
  
INSERT INTO cdr VALUES ("$ 
{caller_id_name}","${caller_id_number}","${destination_number}","$ 
{context}","${s
tart_stamp}","${answer_stamp}","${end_stamp}","${duration}","$ 
{billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}

");
"${caller_id_name}","$ 
{caller_id_number}","${destination_number}","${context}","$ 
{start_stamp}","${answ
er_stamp}","${end_stamp}","${duration}","${billsec}","$ 
{hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","$ 
{read_codec}","${

write_codec}"
"${caller_id_name}","$ 
{caller_id_number}","${destination_number}","${context}","$ 
{start_stamp}","${answer_
stamp}","${end_stamp}","${duration}","${billsec}","$ 
{hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}","$ 
{read_codec}","${wr
ite_codec}","${sip_user_agent}","${call_clientcode}","$ 
{sip_rtp_rxstat}","${sip_rtp_txstat}","${sofia_record_file}"template>
"${caller_id_name}","$ 
{caller_id_number}","${destination_number}","${context}","$ 
{start_stamp}","${answ
er_stamp}","${end_stamp}","${duration}","${billsec}","$ 
{hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","$ 
{read_codec}","${

write_codec}","${sip_user_agent}","${sip_p_rtp_stat}"
"${accountcode}","$ 
{caller_id_number}","${destination_number}","${context}","$ 
{caller_id}","${channel_
name}","${bridge_channel}","${last_app}","${last_arg}","$ 
{start_stamp}","${answer_stamp}","${end_stamp}","${duration}","$ 
{billsec}",

"${hangup_cause}","${amaflags}","${uuid}","${userfield}"
  






call flow is the following:


CALLER => FS =>  CALLED


FS answers the call from CALLER, plays an announcement and bridges  
towards CALLED.



I get different behavior when the call is released by Caller and by  
Called.



Released by Caller:   the CDR is ok having all timestamps

OK CDR:

Outbound LEG =>  
"016659280","016659280","0914392122","public","2009-09-24  
12:02:48","2009-09-24 12:02:54","2009-09-24  
12:03:01","13","7","NORMAL_CLEARING","699cc2d0-a8f1-11de-962a- 
e328afdb9d8d","","","PCMA","PCMA"
Inbound LEG  =>  
"016659280","016659280","05000403","public","2009-09-24  
12:02:27","2009-09-24 12:02:41","2009-09-24  
12:03:01","34","20","NORMAL_CLEARING","5d530192-a8f1-11de-962a- 
e328afdb9d8d","699cc2d0-a8f1-11de-962a-e328afdb9d8d","","PCMA","PCMA"



Released by Called:  the CDR is NOT OK as timestamps are missing


NOT OK CDR:

Inbound LEG =>  
"016659280","016659280","0914392122","public","2009-09-24  
12:05:20","2009-09-24 12:05:30","2009-09-24  
12:05:39","19","9","NORMAL_CLEARING","c479411a-a8f1-11de-962a- 
e328afdb9d8d","","","PCMA","PCMA"
Outbound LEG  
= 
> 
"016659280 
","016659280 
","015000403","public","","","","0","0","NORMAL_CLEARING","b82f2046- 
a8f1-11de-962a-e328afdb9d8d","c479411a-a8f1-11de-962a- 
e328afdb9d8d","","PCMA","PCMA"





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Re: [Freeswitch-users] Status of ubuntu/debian packages.

[Michael Jerris]

I can confirm you should not need the swig dependency at all for  
anything.

Mike

On Sep 24, 2009, at 1:49 PM, William King wrote:

> Hmm... That is interesting... swig is needed I believe only for the
> mod_perl or the esl modules. I'll find out more information and put it
> on the correct package.
>
> I will also update the mod_skypiax config files in the *.install  
> files.
>
> -William King
>
> Dmitry Bely wrote:
>> On Thu, Sep 24, 2009 at 2:14 AM, William King  
>>  wrote:
>>
>>> Sure, post it here and I'll add it in the next build in a few hours.
>>>
>>
>> See attached file.
>>
>> Unfortunately mod_skypiax author did not placed config files
>> (skypiax.conf.xml, skypiax.X.conf) into conf/autoload_configs, so  
>> they
>> are not included into freeswitch-config and should be added manually.
>>
>> BTW, why swig is a dependency for the source package? I recall  
>> Brian's
>> post where he insists that swig is never needed to build Freeswitch.
>>
>>
>>> -William King
>>>
>>> Dmitry Bely wrote:
>>>
 On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle >>> > wrote:


> On Wed, Sep 23, Dmitry Bely wrote:
>
>
>> Can you enable mod_skypiax in your debian package?
>>
>>
> We will be enabling as much as we can cleanly build on debian/ 
> ubuntu.  There will be a lot more to come.  We will be breaking  
> the mods and end points in to different packages so that you can  
> install what you like.  If you have something you would like to  
> see in the package let us know.  Also patches are welcome.
>
>
 Well, mod_skypiax just requires trivial one-line addition to
 debian/rules and debian/freeswitch.install. It builds OK. If the  
 patch
 is required I can post it here.

>>
>> - Dmitry Bely
>>
>> 
>>
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Re: [Freeswitch-users] Freeswitch on embedded device: interesting or not?

[Michael Jerris]

I know of at least one person who has had good luck with small  
applications on arm, in fact there are good working instructions for  
how to cross for arm on the wiki that are known to work.

Mike

On Sep 24, 2009, at 5:34 AM, Cavalera Claudio Luigi wrote:

> Hello guys,
> lately I've been trying to compile Freeswitch for a MIPS architecture.
> With the help of the community I've understood that my target
> architecture was wrong because of limitations in the SDK toolchain's.
> I'm not writing now to get help but to start (I hope) a discussion.
> I would like to understand your points of view about the general  
> idea of
> porting FS on embedded devices.
> I'm not hardware expert at all; someone says that porting FS to any
> appliance which is not x86 based is a loss of time, because ARM and  
> MIPS
> processors just lack computational power, this could be true, but  
> maybe
> it depends on what you expect FS to do on such an embedded  
> architecture.
> We are now all used to the amazing performance of FS on multicores  
> 64bit
> cpus but still the one line description of FS is:
> "FreeSWITCH is an open source telephony platform designed to  
> facilitate
> the creation of voice and chat driven products scaling from a soft- 
> phone
> up to a soft-switch".
> Therefore a scaled down FS could be done, do you think is interesting?
> I'm not speaking here from a technical point of view, I know others  
> have
> already compiled FS for ARM and MIPS and their experience is on the
> wiki.
> Would you consider a scaled down FS only for x86 architectures (e.g.  
> the
> pfSense package or Atom)?
>
> Regards,
> Claudio
>


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Re: [Freeswitch-users] checking subscribed/subscribers to events

[Michael Jerris]

If you need to be able to do granular permissions like that you would  
either need to extend mod_event_socket or write a proxy that handled  
that.

Mike

On Sep 24, 2009, at 5:23 AM, Alberto Escudero wrote:

> Hi,
>
> Is there any simple way to know:
>
> who is subscribed to certain events via ESL?
> check which events i have subscribed during a ESL session?
> control which events can one user subscribe?
> disable the subscription of certain events and not all at the same  
> time?


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Re: [Freeswitch-users] User Creation with DB in Freeswitch

[Michael Jerris]

There are a number of examples out there such as:

http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/intralanman/PHP/fs_curl/

Mike

On Sep 24, 2009, at 7:02 AM, Costa Zikalala wrote:


Hi Gabe

Thanks for you response to this question.
Do you perhaps have a link to an example (or just further detail) to  
what you've descibed below.
I guess one would also use a similar setup to generate dialplans  
from web forms.


Thanks again,
Costa



2009/9/24 Gabriel Gunderson 
On Wed, Sep 23, 2009 at 12:37 PM, Anil Kumar S. R.  
 wrote:
> I didn't get much help for my problem with XML CURL. What I meant  
to say is,
> suppose I want to have some 1 users on freeswitch. Do we have  
to create
> some many xml files in the directory or is there some way in which  
the users

> can be put in the db ?

That's the whole point.  You serve up the XML from *your* web server
using whatever technologies that *you* would like on the back-end.
You'll want to make that XML reply dynamic.

Use php, perl, python, c#, java or whatever other language *you* want.
 Pull the data from MySQL, flat-files, PostgreSQL, MSSQL, LDAP or
whatever *you* want.  Just serve up the right bit of configuration to
FreeSWITCH and you're done.

Good luck.  Spend more time in the docs.  Others have posted the  
links.


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Re: [Freeswitch-users] Subscribing to events in managed C# / .NET

[Michael Jerris]

There are a few other things I can think would be nice additions to  
mod_managed.  Maybe an event handler that does not require a thread to  
be sitting and waiting for events trying in a loop would be nice,  
instead something that is triggered each time there is a certain event  
class triggered.  Also, there has been some interest in doing full  
endpoint modules in mod_managed.  exposing all the state handlers  
in .net like ways and having that all work would be quite interesting,  
but probably requires someone specific actually ready to write a  
module like that to be worthwhile.


Mike

On Sep 24, 2009, at 4:01 AM, Michael Giagnocavo wrote:

Great – hopefully we’ll meet on IRC or the conference sometime on  
Friday. Email me when you’re on.


A few questions I have:

Clarity – I agree with you there, and thanks!

Testability – is this even remotely practical? Looking at our FS  
code plugins, there’s simply no way any amount of test environment  
code would get us to anything testable. We make tons of direct P/ 
Invoke calls, and the whole model for what variables are set when,  
the state machine progression, etc. does not seem like something  
that we can hope to possibly model right. And it’s subject to many  
external influences (all the modules you have loaded in FS). Logging  
is a pretty simple case, sure, we can make it not call FS for  
testing. But in a real app, it just seems that there are way too  
many dependencies, no? Maybe others who have apps written can chime  
in?


Modularity – I agree there are two parts. But, I think they are  
pretty tightly coupled. The FS interface into unmanaged code is done  
via unmanaged code and is really clear: App, Api, ApiBackground. The  
other ways I can think of are FS-specific, such as XML binding  
interface and so on. But those are things we should just add to the  
mod_managed core and be done with. I’m thinking maybe we are talking  
about different things? Can you provide some user stories that we  
want to cover with a pluggable loader/executor/etc.? Thanks for  
putting up with me!


-Michael

From: freeswitch-users-boun...@lists.freeswitch.org  
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of  
Josh Rivers

Sent: Thursday, September 24, 2009 12:32 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Subscribing to events in managed  
C# / .NET




On Wed, Sep 23, 2009 at 7:31 PM, Michael Giagnocavo > wrote:
Right off the bat: there can be tons of cleanup and refactoring, no  
doubt about that. Much of the current code is to satisfy my needs in  
production, which it does very well.


The current base doesn't have anything wrong with it for sure, in  
fact, I learned a good bit about PInvoke. AppDomains, and In-Process  
Remoting in the last week.


My refactoring had the following goals (in no particular order)
 - Testability - I'd really like to see a decent unit test suite on  
the more module so that we can change it with confidence. Also, it's  
been drilled into me that a testable design is a good design.
 - Clarity - Where possible, I extracted blocks of code that served  
a particular purpose so that purpose could be self-documenting in  
the method calls rather than mixed in.
 - Modularity - I wanted to make it easy to remove or add  
alternative behavior to the managed.dll.


I’m a bit hesitant to go too far from the FreeSWITCH core as far as  
architecture goes. For instance, I’m not quite sure why’d we have  
our own managed logging subsystem that allows them to plug in other  
things that aren’t part of FS. Either they should use the FS logging  
system, or use their own such as log4net. Or perhaps I don’t see why  
we’d want this behavior.


I completely agree, with the following caveats:
1) I'd like to see things testable. It's very hard to do isolation  
testing with classes making direct calls out to a static Log class  
that in turn pinvokes out to unmanaged code.
2) I'd like to allow folk to make changes to the default behavior  
(optimally) without recompiling managed.dll.


One thing at issue here is that there are two principal purposes for  
managed.dll. The first is to provide an interface into unmanaged  
code. The second is a module/plugin extensibility framework. The  
first purpose should absolutely provide the thinnest layer possible.  
The second purpose is very likely to need a lot of change and  
adaptation as people come up with development models that they would  
like to follow in using freeswitch. The extensibility framework  
should be mostly managed code, coded to interfaces for mock-ability  
and testabiliy. It should also be able to just push it out of the  
way and hook your own extensibilty framework in instead.
 Going away from the core as far as adding .NET specific features  
(like look at the static ManagedSession.Originate that takes hangup  
delegates, or the “nice” wrapper for Log (Write and WeiteLine, with  
an enum instead of a string) are ke

Re: [Freeswitch-users] Status of ubuntu/debian packages.

[Stephen Crosby]

Thanks William,

This is very helpful.

--Stephen

On Thu, Sep 24, 2009 at 3:58 PM, William King  wrote:

> Alright. I  was able to get the freeswitch project officially on
> launchpad. So here are the new links:
>
> Nightlies:
>
> https://launchpad.net/~freeswitch-drivers/+archive/freeswitch-nightly-drivers
>
> Official releases plus major bug fixes:
> https://launchpad.net/~freeswitch-drivers/+archive/ppa
>
> -William King
>
> Dmitry Bely wrote:
> > On Fri, Sep 25, 2009 at 1:00 AM, Frank Carmickle 
> wrote:
> >
> >> Hello
> >>
> >> On Fri, Sep 25, Dmitry Bely wrote:
> >>
> >>> On Wed, Sep 23, 2009 at 8:08 PM, William King 
> wrote:
> >>>
> >>> Another problem: all music packages in the repository (except 48Khz)
> are empty.
> >>>
> >> If your speaking of the source package then they should be.  If the
> binary package freeswitch-sounds-music-8000 is empty then we have problems.
>  It was working the other day.  Please let me know.
> >>
> >
> > Of course, I meant the binary packages:
> >
> >
> http://ppa.launchpad.net/pbxbuntu-drivers/ppa/ubuntu/pool/main/f/freeswitch-sounds-music/
> >
> > Note their sizes.
> >
> > - Dmitry Bely
> >
> > ___
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> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
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Re: [Freeswitch-users] Transfer hangs.

[Anthony Minessale]

in that case, it's probably a delay in the media stream where the app is
queued when you press the key

try updating to trunk and add the new i flag to the flags param i.e. 1 b ai
transfer::ff-transfer XML public


On Thu, Sep 24, 2009 at 2:04 PM, Harry Vangberg  wrote:

> Not exactly, as I said, if the original B-leg doesn't hang up, it will
> wait 20 second before transfering to the new extension (check the
> timestamps!) - but if the original B leg hangs up, it gets transfered
> to the extension immediately.
>
> Look at this:
>
> 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_async.c:2042
> sofia/external/hemmel...@129.142.224.250 Processing meta digit '1'
> [transfer::ff-transfer XML public]
> 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:813 Send
> signal sofia/external/hemmel...@129.142.224.250 [BREAK]
> 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_bridge.c:228
> sofia/external/hemmel...@129.142.224.250 receive message [UNBRIDGE]
> 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:630 Send
> signal sofia/external/hemmel...@129.142.224.250 [BREAK]
> 2009-09-24 18:29:48.278342 [DEBUG] switch_core_session.c:813 Send
> signal sofia/external/46934488 [BREAK]
> 2009-09-24 18:29:48.298341 [DEBUG] switch_ivr_bridge.c:228
> sofia/external/46934488 receive message [UNBRIDGE]
> 2009-09-24 18:29:48.298341 [DEBUG] switch_core_session.c:630 Send
> signal sofia/external/46934488 [BREAK]
> 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr.c:540
> sofia/external/46934488 Command Execute playback(local_stream://moh)
> EXECUTE sofia/external/46934488 playback(local_stream://moh)
> 2009-09-24 18:29:48.438320 [WARNING] mod_local_stream.c:318 Unknown
> source moh, trying 'default'
> 2009-09-24 18:29:48.438320 [ERR] mod_local_stream.c:327 Unknown source
> default
> 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:231
> sofia/external/46934488 receive message [BRIDGE]
> 2009-09-24 18:29:48.438320 [DEBUG] switch_core_session.c:630 Send
> signal sofia/external/46934488 [BREAK]
> 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:233 Send signal
> sofia/external/hemmel...@129.142.224.250 [BREAK]
> 2009-09-24 18:30:09.111448 [DEBUG] switch_ivr.c:540
> sofia/external/hemmel...@129.142.224.250 Command Execute
> transfer(ff-transfer XML public)
> EXECUTE sofia/external/hemmel...@129.142.224.250 transfer(ff-transfer
> XML public)
>
> >From 18:29:48 to 19:30:09 nothing happens - it's first then it's
> transferred to the new extension, and first after that that the new
> B-leg will even get called.
>
> 2009/9/24 Anthony Minessale :
> > because it's waiting for the other party to answer
> >
> > if you want to hear ringback or music while you are waiting
> > see:
> > http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones
> >
> > specifically transfer_ringback
> >
> >
> > On Thu, Sep 24, 2009 at 1:35 PM, Harry Vangberg 
> wrote:
> >>
> >> Hello
> >>
> >> My setup is this (I've simplified everything, because a lot of my
> >> logic isn't necesarry for showcasing this): A calls in, transfer is
> >> bound as meta app, B is bridged. When the meta app is processed, the
> >> call is transfered to a new extension, which rebridges A. But! After
> >> triggering the meta app, it hangs 20 seconds, until transfering to the
> >> new extension, unless the B-leg hangs up manually.
> >>
> >> It should be noted that I've set dtmf-type=sip-info, as I would like
> >> to bypass media–if there's a better solution to get DTMF events while
> >> bypassing media, please say so, as I know the SIP INFO solution is
> >> kinda havoced.
> >>
> >> This is my dialplan:
> >>
> >> 
> >>  
> >>
> >>
> >>  
> >>
> >>
> >> >> data="sofia/gateway/gw1.fonet.dk/46934488" />
> >>  
> >>
> >>
> >>
> >>  
> >> >> data="sofia/gateway/gw1.fonet.dk/31354228" />
> >>  
> >>
> >>...
> >>   
> >> 
> >>
> >> A full trace of a session with A calling in, B answering, B triggering
> >> meta app, waiting for transfer, and finally bridge to C is attached.
> >>
> >> This is using freeswitch-tr...@14962
> >>
> >> ___
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users@lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_miness...@hotmail.com 
> > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:8...@conference.freeswitch.org 
> > iax:gu...@conference.freeswitch.org/888
> > googletalk:conf+...@conference.freeswitch.org
> > pstn:213-799-1400
> >
> > ___
> > FreeSWITCH-users ma

Re: [Freeswitch-users] Status of ubuntu/debian packages.

[William King]

Alright. I  was able to get the freeswitch project officially on
launchpad. So here are the new links:

Nightlies:
https://launchpad.net/~freeswitch-drivers/+archive/freeswitch-nightly-drivers

Official releases plus major bug fixes:
https://launchpad.net/~freeswitch-drivers/+archive/ppa

-William King

Dmitry Bely wrote:
> On Fri, Sep 25, 2009 at 1:00 AM, Frank Carmickle  wrote:
>   
>> Hello
>>
>> On Fri, Sep 25, Dmitry Bely wrote:
>> 
>>> On Wed, Sep 23, 2009 at 8:08 PM, William King  wrote:
>>>
>>> Another problem: all music packages in the repository (except 48Khz) are 
>>> empty.
>>>   
>> If your speaking of the source package then they should be.  If the binary 
>> package freeswitch-sounds-music-8000 is empty then we have problems.  It was 
>> working the other day.  Please let me know.
>> 
>
> Of course, I meant the binary packages:
>
> http://ppa.launchpad.net/pbxbuntu-drivers/ppa/ubuntu/pool/main/f/freeswitch-sounds-music/
>
> Note their sizes.
>
> - Dmitry Bely
>
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Re: [Freeswitch-users] Polycom MWI Forgetfulness

[digilord]

Here is the requested SIP trace that Anthony wanted.

http://pastebin:freeswi...@pastebin.freeswitch.org/10479

This is for ext 1...@192.168.0.2.

On line 168 the phone thinks there are no messages. 
On line 206 the phone thinks there are no messages.
On line 309 the phone thinks there are messages.
On line 599 the phone thinks there are no messages.
On line 637 the phone thinks there are no messages.



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Re: [Freeswitch-users] Status of ubuntu/debian packages.

[Dmitry Bely]

On Fri, Sep 25, 2009 at 1:00 AM, Frank Carmickle  wrote:
> Hello
>
> On Fri, Sep 25, Dmitry Bely wrote:
>> On Wed, Sep 23, 2009 at 8:08 PM, William King  wrote:
>>
>> Another problem: all music packages in the repository (except 48Khz) are 
>> empty.
>
> If your speaking of the source package then they should be.  If the binary 
> package freeswitch-sounds-music-8000 is empty then we have problems.  It was 
> working the other day.  Please let me know.

Of course, I meant the binary packages:

http://ppa.launchpad.net/pbxbuntu-drivers/ppa/ubuntu/pool/main/f/freeswitch-sounds-music/

Note their sizes.

- Dmitry Bely

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Re: [Freeswitch-users] skill-based ACD

[Andrew Thompson]

On Thu, Sep 24, 2009 at 08:40:16AM -0700, msc wrote:
> On Thu, Sep 24, 2009 at 1:09 AM, Remko Kloosterman 
> wrote:
> 
> >  Hello Michael,
> >
> >
> >
> > Do you still want to follow up on this? I?m having difficulty gathering the
> > old stuff in an understandable form. Also, it looks like the open source ACD
> > Spice Telephony by Andrew Thompson can do just what you might need.
> >
> > I had totally forgotten about Andrew's stuff! Unless people want to build
> their own 100% community/free/DIY version of a skill-based ACD then I say
> let's all play with SpiceCSM and help improve it.

I'd certainly appreciate the feedback (and the kick in the ass to
improve some of the rough spots and documentation).

Andrew

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

[Frank Carmickle]

Hello

On Fri, Sep 25, Dmitry Bely wrote:
> On Wed, Sep 23, 2009 at 8:08 PM, William King  wrote:
> 
> Another problem: all music packages in the repository (except 48Khz) are 
> empty.

If your speaking of the source package then they should be.  If the binary 
package freeswitch-sounds-music-8000 is empty then we have problems.  It was 
working the other day.  Please let me know.

--FC

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

[William King]

You're saying the binary sounds files are empty?

And only the music ones?

-William King

Dmitry Bely wrote:
> On Wed, Sep 23, 2009 at 8:08 PM, William King  wrote:
>
> Another problem: all music packages in the repository (except 48Khz) are 
> empty.
>
> - Dmitry Bely
>
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Re: [Freeswitch-users] Status of ubuntu/debian packages.

[Dmitry Bely]

On Wed, Sep 23, 2009 at 8:08 PM, William King  wrote:

Another problem: all music packages in the repository (except 48Khz) are empty.

- Dmitry Bely

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[Freeswitch-users] instant messaging

[RobertT]

Hi guys!

I'm considering to use SIMPLE protocol for IM in my application, but get a
following error trying to send a message from one registered user to
another:
[ERR] sofia_presence.c:93 Chat proto [sip]
from [1...@xx.xxx.xx.xx]
to [1...@xx.xxx.xx.xx]
11
Invalid Profile xx.xxx.xx.xx

Should presence be enabled in order SIMPLE to work? What additional steps do
I have to complete in order to make presence work in FS besides setting
"manage_presence" param in SIP profile to true?

Best regards, Robert.
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Re: [Freeswitch-users] Polycom MWI Forgetfulness

[digilord]

Brian,
I am using the latest firmware.  Would a solution be to lower the
registration time so that notifies happen more often?  I know that would
increase the amount of traffic on the network but it would keep the
light lit when a user has a message.

DigiLord

On Thu, 2009-09-24 at 13:32 -0500, Brian West wrote:
> It beeps every few min cuz you register and we send you a notify again.
> 
> /b
> 
> On Sep 24, 2009, at 11:43 AM, Chris Burns wrote:
> 
> > This happens with our polycoms as well ... NAT on phone and PBX.  
> > Still haven't
> > had time to look into it so I disabled the sound for new message  
> > waiting ...
> > for now it doesn't keep beeping every few minutes.
> 
> 
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Re: [Freeswitch-users] Transfer hangs.

[Harry Vangberg]

Not exactly, as I said, if the original B-leg doesn't hang up, it will
wait 20 second before transfering to the new extension (check the
timestamps!) - but if the original B leg hangs up, it gets transfered
to the extension immediately.

Look at this:

2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_async.c:2042
sofia/external/hemmel...@129.142.224.250 Processing meta digit '1'
[transfer::ff-transfer XML public]
2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:813 Send
signal sofia/external/hemmel...@129.142.224.250 [BREAK]
2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_bridge.c:228
sofia/external/hemmel...@129.142.224.250 receive message [UNBRIDGE]
2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:630 Send
signal sofia/external/hemmel...@129.142.224.250 [BREAK]
2009-09-24 18:29:48.278342 [DEBUG] switch_core_session.c:813 Send
signal sofia/external/46934488 [BREAK]
2009-09-24 18:29:48.298341 [DEBUG] switch_ivr_bridge.c:228
sofia/external/46934488 receive message [UNBRIDGE]
2009-09-24 18:29:48.298341 [DEBUG] switch_core_session.c:630 Send
signal sofia/external/46934488 [BREAK]
2009-09-24 18:29:48.438320 [DEBUG] switch_ivr.c:540
sofia/external/46934488 Command Execute playback(local_stream://moh)
EXECUTE sofia/external/46934488 playback(local_stream://moh)
2009-09-24 18:29:48.438320 [WARNING] mod_local_stream.c:318 Unknown
source moh, trying 'default'
2009-09-24 18:29:48.438320 [ERR] mod_local_stream.c:327 Unknown source default
2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:231
sofia/external/46934488 receive message [BRIDGE]
2009-09-24 18:29:48.438320 [DEBUG] switch_core_session.c:630 Send
signal sofia/external/46934488 [BREAK]
2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:233 Send signal
sofia/external/hemmel...@129.142.224.250 [BREAK]
2009-09-24 18:30:09.111448 [DEBUG] switch_ivr.c:540
sofia/external/hemmel...@129.142.224.250 Command Execute
transfer(ff-transfer XML public)
EXECUTE sofia/external/hemmel...@129.142.224.250 transfer(ff-transfer
XML public)

>From 18:29:48 to 19:30:09 nothing happens - it's first then it's
transferred to the new extension, and first after that that the new
B-leg will even get called.

2009/9/24 Anthony Minessale :
> because it's waiting for the other party to answer
>
> if you want to hear ringback or music while you are waiting
> see:
> http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones
>
> specifically transfer_ringback
>
>
> On Thu, Sep 24, 2009 at 1:35 PM, Harry Vangberg  wrote:
>>
>> Hello
>>
>> My setup is this (I've simplified everything, because a lot of my
>> logic isn't necesarry for showcasing this): A calls in, transfer is
>> bound as meta app, B is bridged. When the meta app is processed, the
>> call is transfered to a new extension, which rebridges A. But! After
>> triggering the meta app, it hangs 20 seconds, until transfering to the
>> new extension, unless the B-leg hangs up manually.
>>
>> It should be noted that I've set dtmf-type=sip-info, as I would like
>> to bypass media–if there's a better solution to get DTMF events while
>> bypassing media, please say so, as I know the SIP INFO solution is
>> kinda havoced.
>>
>> This is my dialplan:
>>
>> 
>>  
>>
>>    
>>      
>>        
>>        
>>        > data="sofia/gateway/gw1.fonet.dk/46934488" />
>>      
>>    
>>
>>    
>>      
>>        > data="sofia/gateway/gw1.fonet.dk/31354228" />
>>      
>>    
>>    ...
>>   
>> 
>>
>> A full trace of a session with A calling in, B answering, B triggering
>> meta app, waiting for transfer, and finally bridge to C is attached.
>>
>> This is using freeswitch-tr...@14962
>>
>> ___
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>> FreeSWITCH-users@lists.freeswitch.org
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>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_miness...@hotmail.com
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
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>
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org
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Re: [Freeswitch-users] ASR using mod_unimrcp, vlingo

[Anthony Minessale]

if you find the time, can you add that to the wiki too?


On Thu, Sep 24, 2009 at 12:07 PM, Jim Page  wrote:

>  Hi Arsen
>
>
>
> Thanks for your message – it inspired us to do what we should have done in
> the first place, and look at the code. The problem we were having was
> related to grammar files not being available locally. Now we have discovered
> the “builtin:” keyword we are up and running J
>
>
>
> Many thanks
>
> Jim
>
>
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Arsen
> Chaloyan
> *Sent:* 24 September 2009 18:24
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] ASR using mod_unimrcp, vlingo
>
>
>
> Hi Jim,
>
> >From conceptual viewpoint, mod_unimrcp is just an alternate implementation
> of an abstract ASR/TTS interface FreeSWITCH provides.
> Therefore you can use it exactly the same way as other ASR/TTS modules.
> See scripts/javascript/ps_pizza.js in FS tree for a working example.
>
> The only thing you should know and change there is module name
> < var asr = new SpeechDetect(session, "pocketsphinx");
> > var asr = new SpeechDetect(session, "unimrcp");
>
> Typically you can specify any grammar your MRCP server supports.
>  --
>
> *From:* Jim Page 
> *To:* "freeswitch-users@lists.freeswitch.org" <
> freeswitch-users@lists.freeswitch.org>
> *Sent:* Thursday, September 24, 2009 1:25:20 PM
> *Subject:* [Freeswitch-users] ASR using mod_unimrcp, vlingo
>
> Hi All
>
> Has anyone had any experience doing ASR with mod_unimrcp in javascript? In
> particular, how do you deal with grammars? A simple piece of demo code would
> be massively appreciated - the documentation on mod_unimrcp ASR javascript
> bindings is TBD, which I assume means 'to be documented' ... unless it means
> 'to be developed' ...
>
> Also is anyone aware of a vlingo integration for freeswitch?
>
> All the best
> Jim
>
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Re: [Freeswitch-users] Transfer hangs.

[Anthony Minessale]

because it's waiting for the other party to answer

if you want to hear ringback or music while you are waiting
see:
http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones

specifically transfer_ringback


On Thu, Sep 24, 2009 at 1:35 PM, Harry Vangberg  wrote:

> Hello
>
> My setup is this (I've simplified everything, because a lot of my
> logic isn't necesarry for showcasing this): A calls in, transfer is
> bound as meta app, B is bridged. When the meta app is processed, the
> call is transfered to a new extension, which rebridges A. But! After
> triggering the meta app, it hangs 20 seconds, until transfering to the
> new extension, unless the B-leg hangs up manually.
>
> It should be noted that I've set dtmf-type=sip-info, as I would like
> to bypass media–if there's a better solution to get DTMF events while
> bypassing media, please say so, as I know the SIP INFO solution is
> kinda havoced.
>
> This is my dialplan:
>
> 
>  
>
>
>  
>
>
> data="sofia/gateway/gw1.fonet.dk/46934488" />
>  
>
>
>
>  
> data="sofia/gateway/gw1.fonet.dk/31354228" />
>  
>
>...
>   
> 
>
> A full trace of a session with A calling in, B answering, B triggering
> meta app, waiting for transfer, and finally bridge to C is attached.
>
> This is using freeswitch-tr...@14962
>
> ___
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>


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Re: [Freeswitch-users] Polycom MWI Forgetfulness

[Brian West]

It beeps every few min cuz you register and we send you a notify again.

/b

On Sep 24, 2009, at 11:43 AM, Chris Burns wrote:

> This happens with our polycoms as well ... NAT on phone and PBX.  
> Still haven't
> had time to look into it so I disabled the sound for new message  
> waiting ...
> for now it doesn't keep beeping every few minutes.


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Re: [Freeswitch-users] Bind to more than one ethernet interface

[Anthony Minessale]

if you bind the same profile to more than one ip, all your traffic would
come in one ip and out another and cause tremendous confusion.
To see a working example of this problem see asterisk
https://issues.asterisk.org/view.php?id=2358

(note bkw and mikej contribute to this bug)

Here is me reporting a similar issue in IAX (my only negative karma ever on
that site)
https://issues.asterisk.org/view.php?id=7315

moral of the story is, it's unwise to bind multiple ip to a server interface
that uses UDP signalling and the SIP spec requires
a UA to have one specific URL


On Thu, Sep 24, 2009 at 12:42 PM, Michael Collins wrote:

>
>
> On Wed, Sep 23, 2009 at 10:22 PM, Gabriel Gunderson wrote:
>
>> On Wed, Sep 23, 2009 at 10:44 PM, Seven Du  wrote:
>> > It not possible to use 0.0.0.0 for on profile. however, you can create
>> more
>> > sip profiles for each of your interfaces. Search freeswitch-users
>> archievs
>> > then you will find similar topics.
>>
>> It sure would be nice to be able to provide a list of IPs in one
>> profile.  Seems like I run into that need often.  Instead, I end up
>> making two profiles that are mostly the same.
>>
>> Is there a reason why this is the way it is?
>>
>
> A SIP profile is a user agent. By design it services exactly one IP/Port
> combo. If you are finding that you need to have multiple IP's or ports in a
> single SIP profile then that suggests to me that you might be trying to use
> the wrong tool for the job. What are the scenarios where you feel you need a
> single profile to handle multiple IP addresses? My guess is that bkw and the
> gang will have suggestions for an elegant solution that will fit your needs.
>
> -MC
>
>
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Re: [Freeswitch-users] Status of ubuntu/debian packages.

[William King]

Hmm... That is interesting... swig is needed I believe only for the
mod_perl or the esl modules. I'll find out more information and put it
on the correct package.

I will also update the mod_skypiax config files in the *.install files.

-William King

Dmitry Bely wrote:
> On Thu, Sep 24, 2009 at 2:14 AM, William King  wrote:
>   
>> Sure, post it here and I'll add it in the next build in a few hours.
>> 
>
> See attached file.
>
> Unfortunately mod_skypiax author did not placed config files
> (skypiax.conf.xml, skypiax.X.conf) into conf/autoload_configs, so they
> are not included into freeswitch-config and should be added manually.
>
> BTW, why swig is a dependency for the source package? I recall Brian's
> post where he insists that swig is never needed to build Freeswitch.
>
>   
>> -William King
>>
>> Dmitry Bely wrote:
>> 
>>> On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle  
>>> wrote:
>>>
>>>   
 On Wed, Sep 23, Dmitry Bely wrote:

 
> Can you enable mod_skypiax in your debian package?
>
>   
 We will be enabling as much as we can cleanly build on debian/ubuntu.  
 There will be a lot more to come.  We will be breaking the mods and end 
 points in to different packages so that you can install what you like.  If 
 you have something you would like to see in the package let us know.  Also 
 patches are welcome.

 
>>> Well, mod_skypiax just requires trivial one-line addition to
>>> debian/rules and debian/freeswitch.install. It builds OK. If the patch
>>> is required I can post it here.
>>>   
>
> - Dmitry Bely
>   
> 
>
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Re: [Freeswitch-users] Bind to more than one ethernet interface

[Michael Collins]

On Wed, Sep 23, 2009 at 10:22 PM, Gabriel Gunderson  wrote:

> On Wed, Sep 23, 2009 at 10:44 PM, Seven Du  wrote:
> > It not possible to use 0.0.0.0 for on profile. however, you can create
> more
> > sip profiles for each of your interfaces. Search freeswitch-users
> archievs
> > then you will find similar topics.
>
> It sure would be nice to be able to provide a list of IPs in one
> profile.  Seems like I run into that need often.  Instead, I end up
> making two profiles that are mostly the same.
>
> Is there a reason why this is the way it is?
>

A SIP profile is a user agent. By design it services exactly one IP/Port
combo. If you are finding that you need to have multiple IP's or ports in a
single SIP profile then that suggests to me that you might be trying to use
the wrong tool for the job. What are the scenarios where you feel you need a
single profile to handle multiple IP addresses? My guess is that bkw and the
gang will have suggestions for an elegant solution that will fit your needs.

-MC
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Re: [Freeswitch-users] Status of ubuntu/debian packages.

[Dmitry Bely]

On Thu, Sep 24, 2009 at 2:14 AM, William King  wrote:
> Sure, post it here and I'll add it in the next build in a few hours.

See attached file.

Unfortunately mod_skypiax author did not placed config files
(skypiax.conf.xml, skypiax.X.conf) into conf/autoload_configs, so they
are not included into freeswitch-config and should be added manually.

BTW, why swig is a dependency for the source package? I recall Brian's
post where he insists that swig is never needed to build Freeswitch.

> -William King
>
> Dmitry Bely wrote:
>> On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle  
>> wrote:
>>
>>> On Wed, Sep 23, Dmitry Bely wrote:
>>>
 Can you enable mod_skypiax in your debian package?

>>> We will be enabling as much as we can cleanly build on debian/ubuntu.  
>>> There will be a lot more to come.  We will be breaking the mods and end 
>>> points in to different packages so that you can install what you like.  If 
>>> you have something you would like to see in the package let us know.  Also 
>>> patches are welcome.
>>>
>>
>> Well, mod_skypiax just requires trivial one-line addition to
>> debian/rules and debian/freeswitch.install. It builds OK. If the patch
>> is required I can post it here.

- Dmitry Bely


freeswitch.diff
Description: Binary data
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Re: [Freeswitch-users] A little Q931 tool

[Michael Collins]

Excellent, thanks!
-MC

On Thu, Sep 24, 2009 at 4:50 AM, Helmut Kuper wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> I ported my perl based FS-Logfile-Q931-HexDumps-to-pcap script to C
> (linux). It reads FS's logfile (loglevel DEBUG) grabs the Q931 hex dumps
> and puts them in a .pcap file which is directly readable and decodeable
> by Wireshark/tshark.
>
> The big plus here is, that you are able to get an isdn Q931 trace even
> of a call in the past (as long as you have FS in DEBUG loglevel).
>
>
> regards
> Helmut
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFKu10F4tZeNddg3dwRAjV3AKCltpyAhdqB0usC4Z2AFReRNUj/5ACfTmyH
> y6O0PXB/IFzXlSFpPQN13JA=
> =OWEy
> -END PGP SIGNATURE-
>
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Re: [Freeswitch-users] ASR using mod_unimrcp, vlingo

[Jim Page]

Hi Arsen

Thanks for your message - it inspired us to do what we should have done in the 
first place, and look at the code. The problem we were having was related to 
grammar files not being available locally. Now we have discovered the 
"builtin:" keyword we are up and running :)

Many thanks
Jim

From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Arsen 
Chaloyan
Sent: 24 September 2009 18:24
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] ASR using mod_unimrcp, vlingo

Hi Jim,

>From conceptual viewpoint, mod_unimrcp is just an alternate implementation of 
>an abstract ASR/TTS interface FreeSWITCH provides.
Therefore you can use it exactly the same way as other ASR/TTS modules.
See scripts/javascript/ps_pizza.js in FS tree for a working example.

The only thing you should know and change there is module name
< var asr = new SpeechDetect(session, "pocketsphinx");
> var asr = new SpeechDetect(session, "unimrcp");

Typically you can specify any grammar your MRCP server supports.

From: Jim Page 
To: "freeswitch-users@lists.freeswitch.org" 

Sent: Thursday, September 24, 2009 1:25:20 PM
Subject: [Freeswitch-users] ASR using mod_unimrcp, vlingo

Hi All

Has anyone had any experience doing ASR with mod_unimrcp in javascript? In 
particular, how do you deal with grammars? A simple piece of demo code would be 
massively appreciated - the documentation on mod_unimrcp ASR javascript 
bindings is TBD, which I assume means 'to be documented' ... unless it means 
'to be developed' ...

Also is anyone aware of a vlingo integration for freeswitch?

All the best
Jim

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Re: [Freeswitch-users] How to play more than one voice file in play_and_get_digits function

[Diego Viola]

Use phrase macros as Brian said.

On Thu, Sep 24, 2009 at 1:09 PM, Brian West  wrote:

> You can also use phrase macros.  (and no its not just for TTS ;)  )
> /b
>
> On Sep 24, 2009, at 2:02 AM, Michael Collins wrote:
>
> Make sure that mod_file_string is built and loaded and then try the syntax
> that is described here:
> http://wiki.freeswitch.org/wiki/Mod_file_string#Examples
>
> Instead of a comma separated list you can use ! and be sure NOT to put a
> space after the ! because the function delimits the arguments with spaces.
> Try something like this:
>
> $conn->execute("play_and_get_digits", 1 1 1000 #
> /usr/local/freeswitch/en/us/callie/sounds/ivr/please.wav!/usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav
> /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+);
>
> Let us know how it goes.
> -MC
>
>
>
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Re: [Freeswitch-users] Polycom MWI Forgetfulness

[Chris Burns]

This happens with our polycoms as well ... NAT on phone and PBX. Still haven't 
had time to look into it so I disabled the sound for new message waiting ... 
for now it doesn't keep beeping every few minutes.

On September 23, 2009 08:08:39 pm Brian West wrote:
> NO I have never seen it happen what firmware version are you running?
>
> /b
>
> On Sep 23, 2009, at 6:38 PM, Daniel Morrigan wrote:
> > Brian,
> >  It was set for contact.  Would that cause this behavior?
> >
> > Daniel
>
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Re: [Freeswitch-users] ASR using mod_unimrcp, vlingo

[Arsen Chaloyan]

Hi Jim,

>From conceptual viewpoint, mod_unimrcp is just an alternate implementation of 
>an abstract ASR/TTS interface FreeSWITCH provides.
Therefore you can use it exactly the same way as other ASR/TTS modules.
See scripts/javascript/ps_pizza.js in FS tree for a working example.

The only thing you should know and change there is module name
< var asr = new SpeechDetect(session, "pocketsphinx"); 
> var asr = new SpeechDetect(session, "unimrcp");

Typically you can specify any grammar your MRCP server supports.




From: Jim Page 
To: "freeswitch-users@lists.freeswitch.org" 

Sent: Thursday, September 24, 2009 1:25:20 PM
Subject: [Freeswitch-users] ASR using mod_unimrcp, vlingo

Hi All

Has anyone had any experience doing ASR with mod_unimrcp in javascript? In 
particular, how do you deal with grammars? A simple piece of demo code would be 
massively appreciated - the documentation on mod_unimrcp ASR javascript 
bindings is TBD, which I assume means 'to be documented' ... unless it means 
'to be developed' ...

Also is anyone aware of a vlingo integration for freeswitch?

All the best
Jim

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Re: [Freeswitch-users] No ring tone while recording incoming call. Please help.

[msc]

On Wed, Sep 23, 2009 at 7:34 AM, Svetik VOIP  wrote:

> Brian,
>
> Thank yo very much for your reply.
>
> I have tried to add transfer_ringback action, but it did not solve my
> problem.
> Destination phone is ringing, but the person who is calling does not hear
> ringing tone in hte handset.
>
> Is there anything in the logfile that can help you to identify the problem?
>
What kind of system is the calling party connected to? It looks like a 180
is sent out by FS:

2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/
sip:m...@192.168.0.121:5060 entering state [proceeding][180]

At that point the server at the originating side *should* generate pretend
ringing for the calling phone. If that is not happening then you need to see
what's going on at the originating side. Is it a SIP provider?

-MC


>
> Closest I can see is:
> 2009-09-22 17:18:05.02 [DEBUG] switch_ivr_originate.c:1738 Raw Codec
> Activation Success l...@8000hz 1 channel 20ms
> 2009-09-22 17:18:05.02 [DEBUG] switch_ivr_originate.c:1797 Play
> Ringback Tone [%(2000,4000,440.0,480.0)]
> 2009-09-22 17:18:05.447237 [DEBUG] switch_core_io.c:232 sofia/external/
> 4163641...@67.205.74.164 receive message [TRANSCODING_NECESSARY]
> 2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/
> sip:m...@192.168.0.121:5060 entering state [proceeding][180]
> 2009-09-22 17:18:05.463192 [NOTICE] sofia.c:3376 Ring-Ready sofia/internal/
> sip:m...@192.168.0.121:5060!
> 2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel sofia/external/
> 4163641...@67.205.74.164 entering state [terminated][487]
> 2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup sofia/external/
> 4163641...@67.205.74.164 [CS_EXECUTE] [ORIGINATOR_CANCEL]
>
> Thank you,
>
> Igor
>
> >set ringback before record_session and also set transfer_ringback
> >because record_session causes an pre-answer.
> >
> >/b
> >
> >On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote:
> >
> >> Hi,
> >>
> >> I have trouble recording incoming calls with FreeSwitch.
> >>
> >> I have followed the instruction from Misc. Dialplan Tools record
> >> session
> >> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session)
> >> It works well for outgoing calls, but I have the problem with
> >> incoming calls.
> >>
> >> The person who is calling does not hear ring tone, he hears just the
> >> silence until
> >> I pick up the phone. Everything else is working, we can talk,
> >> conversation is recorded.
> >>
> >> Here is a copy of my dialplan for incoming calls
> >> /usr/local/freeswitch/conf/dialplan/public/voipms.xml
> >>
> >> 
> >>
> >>  >> expression="XX">  
> >> 
> >> 
> >>  >> data="RECORD_SOFTWARE=FreeSwitch"/>
> >>  >> data="RECORD_ARTIST=FreeSwitch"/>
> >>  >> data="RECORD_COMMENT=FreeSwitch"/>
> >> 
> >> 
> >> 
> >> 
> >> 
> >> 
> >> 
> >> 
> >
> >
>
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Re: [Freeswitch-users] skill-based ACD

[msc]

On Thu, Sep 24, 2009 at 1:09 AM, Remko Kloosterman wrote:

>  Hello Michael,
>
>
>
> Do you still want to follow up on this? I’m having difficulty gathering the
> old stuff in an understandable form. Also, it looks like the open source ACD
> Spice Telephony by Andrew Thompson can do just what you might need.
>
> I had totally forgotten about Andrew's stuff! Unless people want to build
their own 100% community/free/DIY version of a skill-based ACD then I say
let's all play with SpiceCSM and help improve it.
-MC
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Re: [Freeswitch-users] Intradomain File Transfer - Unsupported Format

[Anthony Minessale]

mime.types file is for http server stuff not SIP
we have never even tried to support file transfer over SIP, it's a feature
request at this point.


On Thu, Sep 24, 2009 at 9:39 AM, David Nembrot wrote:

> Hello list,
>
> I'm currently testing file transfer within the same SIP domain and
> the situation has just got odd! When I send a PDF or MP3 file,
> Freeswitch allows its transfer (meaning SIP Traffic is okay:
> INVITE, 180 Ringing, 200 OK, SEND Transaction, BYE, 200 OK)
> But when I try to send JPEG files, Freeswitch answers to the
> INVITE message with 415 Unsupported Media Type...so there is
> no SEND transaction!
>
> The fact is that I've checked the supported media types list in
> the config file 'mime.types' and I positively found the line:
> 'image/jpeg  jpeg jpg jpe'
>
> There seems to be something wrong in FS, and well actually
> I don't know how to get rid of this Freeswitch's negative reply...
> Could you help me validating such transfer please?
>
> Thanks in advance
>
> David N.
>
>
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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com 
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org 
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[Freeswitch-users] Intradomain File Transfer - Unsupported Format

[David Nembrot]

Hello list,

I'm currently testing file transfer within the same SIP domain and
the situation has just got odd! When I send a PDF or MP3 file,
Freeswitch allows its transfer (meaning SIP Traffic is okay:
INVITE, 180 Ringing, 200 OK, SEND Transaction, BYE, 200 OK)
But when I try to send JPEG files, Freeswitch answers to the
INVITE message with 415 Unsupported Media Type...so there is
no SEND transaction!

The fact is that I've checked the supported media types list in
the config file 'mime.types' and I positively found the line:
'image/jpeg  jpeg jpg jpe'

There seems to be something wrong in FS, and well actually
I don't know how to get rid of this Freeswitch's negative reply...
Could you help me validating such transfer please?

Thanks in advance

David N.


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Re: [Freeswitch-users] Status of ubuntu/debian packages.

[Frank Carmickle]

On Thu, Sep 24, Alberto Escudero wrote:
> Hi,
> 
> Now I seem to reach the webserver. How do i checkout a local copy to run
> the builder?

If you just want to build then you can put a line like 

deb-src http://ppa.launchpad.net/pbxbuntu-drivers/ppa/ubuntu jaunty main

in your sources.list replace jaunty with what ever your running. Then make sure 
you are in a directory where you want the source and

apt-get update -f ; apt-get source freeswitch

HTH
--FC

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Re: [Freeswitch-users] User Creation with DB in Freeswitch

[Costa Zikalala]

Hi Gabe

Thanks for you response to this question.
Do you perhaps have a link to an example (or just further detail) to what
you've descibed below.
I guess one would also use a similar setup to generate dialplans from web
forms.

Thanks again,
Costa



2009/9/24 Gabriel Gunderson 

> On Wed, Sep 23, 2009 at 12:37 PM, Anil Kumar S. R. 
> wrote:
> > I didn't get much help for my problem with XML CURL. What I meant to say
> is,
> > suppose I want to have some 1 users on freeswitch. Do we have to
> create
> > some many xml files in the directory or is there some way in which the
> users
> > can be put in the db ?
>
> That's the whole point.  You serve up the XML from *your* web server
> using whatever technologies that *you* would like on the back-end.
> You'll want to make that XML reply dynamic.
>
> Use php, perl, python, c#, java or whatever other language *you* want.
>  Pull the data from MySQL, flat-files, PostgreSQL, MSSQL, LDAP or
> whatever *you* want.  Just serve up the right bit of configuration to
> FreeSWITCH and you're done.
>
> Good luck.  Spend more time in the docs.  Others have posted the links.
>
>
> Best,
> Gabe
>
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Re: [Freeswitch-users] How to play more than one voice file in play_and_get_digits function

[Brian West]

You can also use phrase macros.  (and no its not just for TTS ;)  )

/b

On Sep 24, 2009, at 2:02 AM, Michael Collins wrote:

Make sure that mod_file_string is built and loaded and then try the  
syntax that is described here:

http://wiki.freeswitch.org/wiki/Mod_file_string#Examples

Instead of a comma separated list you can use ! and be sure NOT to  
put a space after the ! because the function delimits the arguments  
with spaces. Try something like this:


$conn->execute("play_and_get_digits", 1 1 1000 # /usr/local/ 
freeswitch/en/us/callie/sounds/ivr/please.wav!/usr/local/freeswitch/ 
en/us/callie/sounds/ivr/press-1.wav  /usr/loca/freeswitch/en/us/ 
callie/sounds/ivr/invalid.wav res \\d+);


Let us know how it goes.
-MC


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Re: [Freeswitch-users] Looking For A-Z

[Brian West]

This belongs on freeswitch-biz

/b

On Sep 24, 2009, at 5:09 AM, Dome Charoenyost wrote:

> Dear Sir,
> I'm looking for A-Z price  and quality should be same
> http://voicetrading.com. Now i use http://voicetrading.com it's good
> quality but very bad support. some time i can payment by credit card,
> paypal some time can't  i don't know why.
>
> BG
>
> Dome C.
>
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[Freeswitch-users] A little Q931 tool

[Helmut Kuper]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hello,

I ported my perl based FS-Logfile-Q931-HexDumps-to-pcap script to C
(linux). It reads FS's logfile (loglevel DEBUG) grabs the Q931 hex dumps
and puts them in a .pcap file which is directly readable and decodeable
by Wireshark/tshark.

The big plus here is, that you are able to get an isdn Q931 trace even
of a call in the past (as long as you have FS in DEBUG loglevel).


regards
Helmut
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFKu10F4tZeNddg3dwRAjV3AKCltpyAhdqB0usC4Z2AFReRNUj/5ACfTmyH
y6O0PXB/IFzXlSFpPQN13JA=
=OWEy
-END PGP SIGNATURE-

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

[Alberto Escudero]

Hi,

Now I seem to reach the webserver. How do i checkout a local copy to run
the builder?

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

> It seems I had a port forwarded incorrectly for the external access to
> the git web interface. here it is again:
>
> http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/
>
> I've tested it to work now.
>
> -William King
>
> Hadley Rich wrote:
>> On Thu, 24 Sep 2009 09:18:23 Frank Carmickle wrote:
>>
>>> Currently it's /opt/freeswitch.  I would like to see it move to FHS
>>> correct
>>>  locations for inclusion in to debian/ubuntu.  This is the next bit
>>> that I
>>>  will be working on.
>>>
>>
>> Yeah, the FHS stuff was the bit that I got a little stuck on a while
>> back.
>>
>>
>>>  Of course we also hope that the debian voip team will pick it
>>>  up once we've cleaned it up.
>>>
>>
>> Sounds good.
>>
>>
>>> I am not an ubuntu guy so I can't speak to that.  I would say that most
>>> of
>>>  the licenses of the included packages would allow for inclusion in
>>> debian
>>>  main.  Things like the cepstral support would have to go in to
>>> contrib.
>>>
>>
>> Gotcha, multiverse is for "not free" software, so anything that can go
>> into
>> main in Debian could go into universe in Ubuntu.
>>
>> hads
>>
>
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[Freeswitch-users] mod_cdr_csv missing timestamps in A-LEG

[Tihomir Culjaga]

hello,

i'm on latest trunk and for some reason i cannot get timestamps dumped in my
cdrs. I use mod_cdr_csv with default settings plus i enabled to get both a
and b legs dumped.


cdr_csv.conf.xml:


  








  
  
INSERT INTO cdr VALUES
("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${s
tart_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}",
"${accountcode}
");
"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answ
er_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${
write_codec}"
"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_
stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}",
"${accountcode}","${read_codec}","${wr
ite_codec}","${sip_user_agent}","${call_clientcode}","${sip_rtp_rxstat}","${sip_rtp_txstat}","${sofia_record_file}"
"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answ
er_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${
write_codec}","${sip_user_agent}","${sip_p_rtp_stat}"
"${accountcode}","${caller_id_number}","${destination_number}","${context}","${caller_id}","${channel_
name}","${bridge_channel}","${last_app}","${last_arg}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}",
"${hangup_cause}","${amaflags}","${uuid}","${userfield}"
  






call flow is the following:


CALLER => FS =>  CALLED


FS answers the call from CALLER, plays an announcement and bridges towards
CALLED.


I get different behavior when the call is released by Caller and by Called.


Released by Caller:   the CDR is ok having all timestamps

OK CDR:

Outbound LEG => "016659280","016659280","0914392122","public","2009-09-24
12:02:48","2009-09-24 12:02:54","2009-09-24
12:03:01","13","7","NORMAL_CLEARING","699cc2d0-a8f1-11de-962a-e328afdb9d8d","","","PCMA","PCMA"
Inbound LEG  => "016659280","016659280","05000403","public","2009-09-24
12:02:27","2009-09-24 12:02:41","2009-09-24
12:03:01","34","20","NORMAL_CLEARING","5d530192-a8f1-11de-962a-e328afdb9d8d","699cc2d0-a8f1-11de-962a-e328afdb9d8d","","PCMA","PCMA"


Released by Called:  the CDR is NOT OK as timestamps are missing


NOT OK CDR:

Inbound LEG => "016659280","016659280","0914392122","public","2009-09-24
12:05:20","2009-09-24 12:05:30","2009-09-24
12:05:39","19","9","NORMAL_CLEARING","c479411a-a8f1-11de-962a-e328afdb9d8d","","","PCMA","PCMA"
Outbound LEG =>"016659280","016659280","015000403","public",*"","","",*
"0","0","NORMAL_CLEARING","b82f2046-a8f1-11de-962a-e328afdb9d8d","c479411a-a8f1-11de-962a-e328afdb9d8d","","PCMA","PCMA"





What can be wrong?
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[Freeswitch-users] Looking For A-Z

[Dome Charoenyost]

Dear Sir,
 I'm looking for A-Z price  and quality should be same
http://voicetrading.com. Now i use http://voicetrading.com it's good
quality but very bad support. some time i can payment by credit card,
paypal some time can't  i don't know why.

BG

Dome C.

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[Freeswitch-users] Freeswitch on embedded device: interesting or not?

[Cavalera Claudio Luigi]

Hello guys,
lately I've been trying to compile Freeswitch for a MIPS architecture.
With the help of the community I've understood that my target
architecture was wrong because of limitations in the SDK toolchain's.
I'm not writing now to get help but to start (I hope) a discussion.
I would like to understand your points of view about the general idea of
porting FS on embedded devices.
I'm not hardware expert at all; someone says that porting FS to any
appliance which is not x86 based is a loss of time, because ARM and MIPS
processors just lack computational power, this could be true, but maybe
it depends on what you expect FS to do on such an embedded architecture.
We are now all used to the amazing performance of FS on multicores 64bit
cpus but still the one line description of FS is:
"FreeSWITCH is an open source telephony platform designed to facilitate
the creation of voice and chat driven products scaling from a soft-phone
up to a soft-switch".
Therefore a scaled down FS could be done, do you think is interesting?
I'm not speaking here from a technical point of view, I know others have
already compiled FS for ARM and MIPS and their experience is on the
wiki.
Would you consider a scaled down FS only for x86 architectures (e.g. the
pfSense package or Atom)?

Regards,
Claudio


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e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 
Codice in materia di protezione dei dati personali). Se avete ricevuto questa 
comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e 
di distruggere il messaggio originale e ogni file allegato senza farne copia 
alcuna o riprodurne in alcun modo il contenuto. 

This e-mail and its attachments are intended for the addressee(s) only and are 
confidential and/or may contain legally privileged information. If you have 
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may take no action based on it, and you may not copy or show it to anyone; 
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[Freeswitch-users] checking subscribed/subscribers to events

[Alberto Escudero]

Hi,

Is there any simple way to know:

who is subscribed to certain events via ESL?
check which events i have subscribed during a ESL session?
control which events can one user subscribe?
disable the subscription of certain events and not all at the same time?

/aep

-- 
Stopping junk mailers is good for the environment




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Re: [Freeswitch-users] Bind to more than one ethernet interface

[Alberto Escudero]

If I am correct you need to create a sip profile per interface and
hardcode/set the IP address of each interface correctly in the SIP RTP
fields of the profile.

Then you need to set carefully the correct NAT and auth options for each
profile

/aep


-- 
Stopping junk mailers is good for the environment

> Hello,
>
>   I am trying to run FreeSwitch on a machine which has more than one
> interface, all of them should be used for SIP. The FreeSwitch binds only
> to
> the first one. I tried setting bind_server_ip to either "auto" or 0.0.0.0
> but it doesn't help.
>
> Any idea what to do?
>
> Thanks! _Yehavi:
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[Freeswitch-users] ASR using mod_unimrcp, vlingo

[Jim Page]

Hi All

Has anyone had any experience doing ASR with mod_unimrcp in javascript? In 
particular, how do you deal with grammars? A simple piece of demo code would be 
massively appreciated - the documentation on mod_unimrcp ASR javascript 
bindings is TBD, which I assume means 'to be documented' ... unless it means 
'to be developed' ...

Also is anyone aware of a vlingo integration for freeswitch?

All the best
Jim

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Re: [Freeswitch-users] skill-based ACD

[Remko Kloosterman]

Hello Michael,

 

Do you still want to follow up on this? I'm having difficulty gathering
the old stuff in an understandable form. Also, it looks like the open
source ACD Spice Telephony by Andrew Thompson can do just what you might
need. 

 

Remko

 

Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Michael
Collins
Verzonden: donderdag 17 september 2009 22:58
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] skill-based ACD

 

 

On Thu, Sep 17, 2009 at 12:13 PM, Remko Kloosterman
 wrote:

I have been working on several voice projects in the past with ACD
features, mostly based on TDM technology. It's all commercial stuff, but
I have the experience and I am willing to share that. If anyone wishes
to start such a development I'm sure I can dig up a functional model and
help with the design. 

 

I would like to see the functional model. That sounds interesting. We
could take it from there. Perhaps the FS community will have a few
members willing to help out.
-MC

 

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Re: [Freeswitch-users] Subscribing to events in managed C# / .NET

[Michael Giagnocavo]

Great - hopefully we'll meet on IRC or the conference sometime on Friday. Email 
me when you're on.

A few questions I have:

Clarity - I agree with you there, and thanks!

Testability - is this even remotely practical? Looking at our FS code plugins, 
there's simply no way any amount of test environment code would get us to 
anything testable. We make tons of direct P/Invoke calls, and the whole model 
for what variables are set when, the state machine progression, etc. does not 
seem like something that we can hope to possibly model right. And it's subject 
to many external influences (all the modules you have loaded in FS). Logging is 
a pretty simple case, sure, we can make it not call FS for testing. But in a 
real app, it just seems that there are way too many dependencies, no? Maybe 
others who have apps written can chime in?

Modularity - I agree there are two parts. But, I think they are pretty tightly 
coupled. The FS interface into unmanaged code is done via unmanaged code and is 
really clear: App, Api, ApiBackground. The other ways I can think of are 
FS-specific, such as XML binding interface and so on. But those are things we 
should just add to the mod_managed core and be done with. I'm thinking maybe we 
are talking about different things? Can you provide some user stories that we 
want to cover with a pluggable loader/executor/etc.? Thanks for putting up with 
me!

-Michael

From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Josh Rivers
Sent: Thursday, September 24, 2009 12:32 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET


On Wed, Sep 23, 2009 at 7:31 PM, Michael Giagnocavo 
mailto:m...@giagnocavo.net>> wrote:

Right off the bat: there can be tons of cleanup and refactoring, no doubt about 
that. Much of the current code is to satisfy my needs in production, which it 
does very well.
The current base doesn't have anything wrong with it for sure, in fact, I 
learned a good bit about PInvoke. AppDomains, and In-Process Remoting in the 
last week.

My refactoring had the following goals (in no particular order)
 - Testability - I'd really like to see a decent unit test suite on the more 
module so that we can change it with confidence. Also, it's been drilled into 
me that a testable design is a good design.
 - Clarity - Where possible, I extracted blocks of code that served a 
particular purpose so that purpose could be self-documenting in the method 
calls rather than mixed in.
 - Modularity - I wanted to make it easy to remove or add alternative behavior 
to the managed.dll.


I'm a bit hesitant to go too far from the FreeSWITCH core as far as 
architecture goes. For instance, I'm not quite sure why'd we have our own 
managed logging subsystem that allows them to plug in other things that aren't 
part of FS. Either they should use the FS logging system, or use their own such 
as log4net. Or perhaps I don't see why we'd want this behavior.
I completely agree, with the following caveats:
1) I'd like to see things testable. It's very hard to do isolation testing with 
classes making direct calls out to a static Log class that in turn pinvokes out 
to unmanaged code.
2) I'd like to allow folk to make changes to the default behavior (optimally) 
without recompiling managed.dll.

One thing at issue here is that there are two principal purposes for 
managed.dll. The first is to provide an interface into unmanaged code. The 
second is a module/plugin extensibility framework. The first purpose should 
absolutely provide the thinnest layer possible. The second purpose is very 
likely to need a lot of change and adaptation as people come up with 
development models that they would like to follow in using freeswitch. The 
extensibility framework should be mostly managed code, coded to interfaces for 
mock-ability and testabiliy. It should also be able to just push it out of the 
way and hook your own extensibilty framework in instead.

 Going away from the core as far as adding .NET specific features (like look at 
the static ManagedSession.Originate that takes hangup delegates, or the "nice" 
wrapper for Log (Write and WeiteLine, with an enum instead of a string) are 
keeping close to the core, just adding a tiny bit of API cleanup. FreeSWITCH 
exposes a lot of strings, and while maybe that's important for some languages, 
.NET users are going to expect stronger typing. But I don't think these types 
of things get people away from FreeSWITCH much.
No disagreement here. I would like to see these things made available by 
interface rather than concrete implementation. It's currently not possible to 
test a plugin without loading it into FS. That precludes automated testing, and 
leaves a pretty big round-trip to test a tweak. I'm a sloppy coder too, so I'm 
always introducing interesting regressions, and that's why I like doing my 
testing without having to b

Re: [Freeswitch-users] mod_fax not working

[Mathieu Parent]

On Wed, Sep 23, 2009 at 11:44 PM,   wrote:
> Hi Mathieu

Hi

> Does this mean you are able to use email-to-fax?
> If yes, would yes would you care to briefly describe how you configured that.

Not yet, but I plan to do so. Il will post my setup in FS wiki.

I first have to solve the receiving problem (fax2mail is more
important than mail2fax for us).

> Thanks
> MC
>

Mathieu Parent

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Re: [Freeswitch-users] How to play more than one voice file in play_and_get_digits function

[Michael Collins]

Make sure that mod_file_string is built and loaded and then try the syntax
that is described here:
http://wiki.freeswitch.org/wiki/Mod_file_string#Examples

Instead of a comma separated list you can use ! and be sure NOT to put a
space after the ! because the function delimits the arguments with spaces.
Try something like this:

$conn->execute("play_and_get_digits", 1 1 1000 #
/usr/local/freeswitch/en/us/callie/sounds/ivr/please.wav!/usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav
/usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+);

Let us know how it goes.
-MC


On Wed, Sep 23, 2009 at 11:38 PM, velusamy velu wrote:

> Dear All,
>
>I am in the process of doing IVR development on FreeSWITCH. I am
> having doubt in the play_and_get_digits application. I am  using Perl
> language for handling IVR.
>
>  How can I play more than one sound file in play_get_digits
> application?
>  For an example,
>
> $conn->execute("play_and_get_digit", 1 1 1000 #
> /usr/local/freeswitch/en/us/callie/sounds/ivr/please.wav ,
> /usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav
> /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+);
>
> Is the above statement is right?  Can I use more than one file in it?
>
> OR
>
> Can I use the play_get_digits as following?
>
> $conn->execute("play_and_get_digit", 1 1 1000 #
> /usr/local/freeswitch/en/us/callie/sounds/ivr/please.av
> /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+);
>
> $conn->execute("play_and_get_digit", 1 1 1000 #
> /usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav
> /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+);
>
> In the above statements how can I assure that the first  application or
> second application is executed?
>
> When the digit is get while playing the first application the second
> application should not be played. How Can I do that?
>
> Is this my understanding wrong?
> Correct me If I am wrong?
>
>
> Please help me?
>
>
>
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