Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?

[Jason White]

Mitul Limbani  wrote:
>  Another idea would be to write simple rsync method, and post a page
> on the same on the Wiki so all those people who have their own server
> and willing to spare some bandwidth can mirror the entire Wiki
> locally.

MediaWiki uses a database, as I understand it. However, there might be a way
to have it write out all of the content to a file of some sort.


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Re: [Freeswitch-users] Skypiax as a windows service

[Giovanni Maruzzelli]

On Fri, Apr 17, 2009 at 12:58 AM, UV  wrote:
> Ok, I think I know where's the confusion here. Let me clarify:
> 1. FS run beautifully as a service - that's why I assumed it should work.
> 2. Skype client runs as a service very well too.
> 3. When running FS as a service with Skypiax (hence Skypiax as a service),
> Skypiax doesn't seem to find the SkypeAPI.

Why mod_skypiax do not find the API? I know for sure that other
services can access the API on Skype clients running as services. So
mod_skypiax is encountering some specific problem.

I will explore into this one and I'll be back to you.

>
> In the Wiki page
> http://wiki.freeswitch.org/wiki/Skypiax#Running_Skypiax_on_Windows_as_a_Serv
> ice it's says that Running Skypiax on Windows as a Service is "Not yet
> written" therefore I assumed it's a known limitation.
>
> Are you saying it isn't?

Was just the documentation "not yet written", I corrected the wiki
page, now reads: "This part of the How To documentation has not yet
been written. Please, feel free to contribute."

>
> Anyway, the farming solution you suggested should solve the problem - I'd
> assume.

As per the previous Anthony's post, you can use FS itself as a farming solution.

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Re: [Freeswitch-users] Skypiax as a windows service

[Giovanni Maruzzelli]

On Fri, Apr 17, 2009 at 12:30 AM, Anthony Minessale
 wrote:
> are you planning on just signaling on TCP or both audio and signalling
> cos realtime audio over TCP kinda stinks.
>
> you may find that just running FS as the farm and calling to it with sip is
> more or less the same idea with no work ;)
>

Hi Anthony,
yes, TCP is not the best for audio. But it's the only way to route
audio from/to the Skype client instance. I mean, it's the only way the
Skype client allows you to access its audio streams.

This is the current situation:
1) mod_skypiax use native signaling (Windows messages, or X events) to
interact with the Skype client through the Skype API.
2) one of the Skype API commands allows for telling to the Skype
client: "please, use this TCP port for audio in, and that TCP port for
audio out, instead of the soundcard".
3) the TCP ports must be on the local IP interface (127.0.0.1)
4) mod_skypiax and the Skype client(s) exchange audio samples through
TCP on the local machine, while signaling is platform native

I would like to have the Skype client instances on another machine,
for security and stability purposes (I'm not trusting consumer grade
Skype client to run on production main FS server).

That's why I was writing the "farming client", for rerouting both the
signaling commands and the audio streams back and forth between two
separate machines.

Now I understand what you wrote: I can use FS itself (with
mod_skypiax) as a "farming client", and connect with the "main" FS via
SIP. So I can achieve the original aim of having a separate machine(s)
with the Skype instances. Obviously, if that's a requirements, I can
optimize the footprint of the "farming client FS" loading only the
modules needed for SIP-Skype interaction.

Thanks a lot Anthony, this cuts the Gordian knot and spare me lots of
pathetic efforts :-)

UV, is this solution practical for you?

Sincerely,

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






>
> On Thu, Apr 16, 2009 at 10:09 AM, Giovanni Maruzzelli 
> wrote:
>>
>> EG: in the "farm out" scenario there will be FS talking via TCP to a
>> "farm client" (on local machine or remote). The "farm client" talks
>> with Skype client instances running on the same machine the "farm
>> client" is running on.
>>
>> On Thu, Apr 16, 2009 at 1:47 PM, UV  wrote:
>> > Decoupling the Skyiax from FS will solve the problem as I assume it'll
>> > use
>> > TCP/IP (winsock) to interface with FS - therefore, I can run it still on
>> > the
>> > same machine but two separate sessions.
>>
>> yes, it uses TCP for this. So you would end up with FS (with Skypiax
>> module) running on RDP while the Skype client instances are running as
>> services, on the same machine (or in different machines). FS will talk
>> to Skype client instances via TCP.
>> Is this acceptable to you?
>>
>> Other question: why not running FS as a service too? If you run FS as
>> a service and Skype clients as services, all things would works? Why
>> you want to use RDP for? (sorry for the silly questions, I just want
>> to understand better).
>>
>> > However, I think getting the Skypiax
>> > to work as a service will be more beneficial regardless if it's
>> > decoupled or
>> > not.
>>
>> What do you mean? I believe that Skypiax (as an FS module) works when
>> FS is run as service. Your problem seems to me that you cannot run
>> Skype instances under RDP because they cannot access the sound device.
>> Is this correct?
>>
>> gm
>>
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>
>
> --
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>
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Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?

[Diego Viola]

The rsync idea sounds better.

Regards,

Diego

On Fri, Apr 17, 2009 at 4:03 AM, Mitul Limbani  wrote:

> Another idea would be to write simple rsync method, and post a page on the
> same on the Wiki so all those people who have their own server and willing
> to spare some bandwidth can mirror the entire Wiki locally.
>
> I am willing to provide a mirror for the existing wiki on a rsync update
> daily for the Indian / Asian crowd on my server.
>
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions,
> The Enterprise Linux Company (TM),
> www.enterux.com
> +91-9820332422
>
> On Thu 16/04/09 15:50 , Will Boyce m...@willboyce.com sent:
>
> It may be worth looking at
> http://www.mediawiki.org/wiki/Extension:Pdf_Export or
> http://www.mediawiki.org/wiki/Extension:Pdf_Book
>
> --
> Regards,
>
> Will Boyce
> tel: 07933 515 987
> url: http://willboyce.com
>
> - "Michael Collins" wrote:
> | From: "Michael Collins"
> | To: freeswitch-users@lists.freeswitch.org
> | Sent: Wednesday, 15 April, 2009 18:12:26 GMT +00:00 GMT Britain, Ireland,
> Portugal
> | Subject: Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF
> ?
> |
> | Unfortunately this isn't being maintained and Bret didn't give his script
> to any of us. If anyone out there is familiar with converting wiki pages to
> PDF and is willing to pick this up then by all means contact me off list and
> we'll discuss it.
> |
> | -MC
> |
> | | On Wed, Apr 15, 2009 at 12:22 AM, Mitul Limbani wrote:
> |
>>
>> Hello there,
>> |
>> | In my previous encounter with FreeSwitch, I had found that Bret had
>> posted on the Mailing List somewhere about availability of the entire
>> FreeSwitch Wiki Documentation on a single PDF, this is useful coz at the
>> offset apart from Wiki there is no other offline media to learn it.
>> |
>> | Is the same PDF available looking at the growth of Wiki pages and the
>> updation.
>> |
>> | I look forward to hear from you guys,
>> |
>> | Thanks & Regards,
>> | Mitul Limbani,
>> | Founder & CEO,
>> | Enterux Solutions,
>> | The Enterprise Linux Company (TM),
>> | www.enterux.com
>> | +91-9820332422
>> |
>> |
>> --
>> Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/
>> | ___
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>> http://lists.freeswitch.org/mailman/options/freeswitch-users
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>> |
>> |
>
>
> |
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Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?

[Mitul Limbani]

 Another idea would be to write simple rsync method, and post a page
on the same on the Wiki so all those people who have their own server
and willing to spare some bandwidth can mirror the entire Wiki
locally.
 I am willing to provide a mirror for the existing wiki on a rsync
update daily for the Indian / Asian crowd on my server.
 Thanks & Regards, 
 Mitul Limbani, 
 Founder & CEO, 
 Enterux Solutions, 
 The Enterprise Linux Company (TM), 
 www.enterux.com 
 +91-9820332422 
 On Thu 16/04/09 15:50 , Will Boyce m...@willboyce.com sent:
 It may be worth looking at
http://www.mediawiki.org/wiki/Extension:Pdf_Export [1] or
http://www.mediawiki.org/wiki/Extension:Pdf_Book [2]
 -- 
 Regards, 
 Will Boyce  
 tel: 07933 515 987 
 url: http://willboyce.com [3] 
 - "Michael Collins"  wrote: 
 | From: "Michael Collins" 
 | To: freeswitch-users@lists.freeswitch.org
 | Sent: Wednesday, 15 April, 2009 18:12:26 GMT +00:00 GMT Britain,
Ireland, Portugal
 | Subject: Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on
Single PDF ?
 |
 | Unfortunately this isn't being maintained and Bret didn't give his
script to any of us. If anyone out there is familiar with converting
wiki pages to PDF and is willing to pick this up then by all means
contact me off list and we'll discuss it.
 |  
 | -MC
 | 
 | | On Wed, Apr 15, 2009 at 12:22 AM, Mitul Limbani  wrote:
 |  Hello there,
 |  
 |  In my previous encounter with FreeSwitch, I had found that Bret
had posted on the Mailing List somewhere about availability of the
entire FreeSwitch Wiki Documentation on a single PDF, this is useful
coz at the offset apart from Wiki there is no other offline media to
learn it.
 |  
 |  Is the same PDF available looking at the growth of Wiki pages and
the updation.
 |  
 |  I look forward to hear from you guys,
 |  
 |  Thanks & Regards, 
 |  Mitul Limbani, 
 |  Founder & CEO, 
 |  Enterux Solutions, 
 |  The Enterprise Linux Company (TM), 
 |  www.enterux.com [5] 
 |  +91-9820332422 
 |  
 | 
-
Msg sent via Enterux Enterprise Email Server :
http://www.enterux.com/ [6]
 | ___
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 |  
 |  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8]
 | 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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 | 
 |  
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-
Msg sent via Enterux Enterprise Email Server :
http://www.enterux.com/

Links:
--
[1] http://www.mediawiki.org/wiki/Extension:Pdf_Export
[2] http://www.mediawiki.org/wiki/Extension:Pdf_Book
[3] http://willboyce.com
[5] http://www.enterux.com
[6] http://www.enterux.com/
[8] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[9] http://lists.freeswitch.org/mailman/options/freeswitch-users
[10] http://www.freeswitch.org
[11] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[12] http://lists.freeswitch.org/mailman/options/freeswitch-users
[13] http://www.freeswitch.org
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Re: [Freeswitch-users] Call bridge in free switch

[Prabhuram Mohan]

Thanks Brain/ Dave,

I ran the modified command as follows

originate sofia/default/1...@192.168.1.102 &conference(3085-192.168.1.102)

this time fs is able to create channel but  am getting a different error :
"sofia.c:3845 Cannot Blind Tranfer one legged call"

Prabhu

On Thu, Apr 16, 2009 at 8:58 PM, David Knell  wrote:

> Take out the brackets -
> originate sofia/profile/1001...
> (and you might want to replace profile with the name of the profile to
> use)
>
> There's documentation here which might help:
> http://wiki.freeswitch.org/wiki/Mod_commands#originate
>
> --Dave
>
> > Hi Mike,
> >
> > I tried the following command per ur advice.. but getting the error
> > CHAN_NOT_IMPLEMENTED
> >
> > originate (sofia/profile/1...@192.168.1.108) &
> > conference(3085-192.168.1.102);
> >
> >
> > freeswi...@internal> originate (sofia/profile/1...@192.168.1.102) &
> > conference(3085-192.168.1.102);
> > -ERR CHAN_NOT_IMPLEMENTED
> >
> > freeswi...@internal> 2009-04-16 20:28:30 [ERR]
> > switch_core_session.c:303 switch_core_session_outgoing_channel() Could
> > not locate channel type (sofia
> > 2009-04-16 20:28:30 [ERR] switch_ivr_originate.c:1486
> > switch_ivr_originate() Cannot create outgoing channel of type [(sofia]
> > cause: [CHAN_NOT_IMPLEMENTED]
> > 2009-04-16 20:28:30 [DEBUG] switch_ivr_originate.c:2084
> > switch_ivr_originate() Originate Resulted in Error Cause: 66
> > [CHAN_NOT_IMPLEMENTED]
> >
> > Thanks
> > prabhu
> >
> > On Thu, Apr 16, 2009 at 4:29 PM, Michael Collins 
> > wrote:
> > Do you need to monitor the possible failure of one of these
> > calls? Just curious. You can call them individually and drop
> > them into a conference right at the FS cmd line:
> >
> > originate (sofia/profile/1...@192.168.1.108) &
> > conference(myconfname);
> > originate (sofia/profile/1...@192.168.1.108) &
> > conference(myconfname);
> > originate (sofia/profile/1...@192.168.1.108) &
> > conference(myconfname);
> >
> > You can control the conference behavior with numerous options.
> > See http://wiki.freeswitch.org/wiki/Mod_conference for lots of
> > great information.
> >
> > -MC
> >
> >
> > On Thu, Apr 16, 2009 at 1:09 AM, Prabhuram Mohan
> >  wrote:
> >
> >
> > Hello,
> >
> > I am trying to find a way to this through
> > fs_cli
> > 1) call out to ClientA (1...@192.168.1.108),
> > ClientB (1...@192.168.1.108) & ClientC
> > (1...@192.168.1.108)
> > 2) Bridge all the 3 legs together into one
> > call
> >
> > Thanks
> > Prabhu
> >
> >
> >
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Re: [Freeswitch-users] Call bridge in free switch

[David Knell]

Take out the brackets -
originate sofia/profile/1001...
(and you might want to replace profile with the name of the profile to
use)

There's documentation here which might help:
http://wiki.freeswitch.org/wiki/Mod_commands#originate

--Dave

> Hi Mike,
> 
> I tried the following command per ur advice.. but getting the error
> CHAN_NOT_IMPLEMENTED
> 
> originate (sofia/profile/1...@192.168.1.108) &
> conference(3085-192.168.1.102);
> 
> 
> freeswi...@internal> originate (sofia/profile/1...@192.168.1.102) &
> conference(3085-192.168.1.102);
> -ERR CHAN_NOT_IMPLEMENTED
> 
> freeswi...@internal> 2009-04-16 20:28:30 [ERR]
> switch_core_session.c:303 switch_core_session_outgoing_channel() Could
> not locate channel type (sofia
> 2009-04-16 20:28:30 [ERR] switch_ivr_originate.c:1486
> switch_ivr_originate() Cannot create outgoing channel of type [(sofia]
> cause: [CHAN_NOT_IMPLEMENTED]
> 2009-04-16 20:28:30 [DEBUG] switch_ivr_originate.c:2084
> switch_ivr_originate() Originate Resulted in Error Cause: 66
> [CHAN_NOT_IMPLEMENTED]
> 
> Thanks
> prabhu
> 
> On Thu, Apr 16, 2009 at 4:29 PM, Michael Collins 
> wrote:
> Do you need to monitor the possible failure of one of these
> calls? Just curious. You can call them individually and drop
> them into a conference right at the FS cmd line:
> 
> originate (sofia/profile/1...@192.168.1.108) &
> conference(myconfname);
> originate (sofia/profile/1...@192.168.1.108) &
> conference(myconfname);
> originate (sofia/profile/1...@192.168.1.108) &
> conference(myconfname);
> 
> You can control the conference behavior with numerous options.
> See http://wiki.freeswitch.org/wiki/Mod_conference for lots of
> great information.
> 
> -MC
> 
> 
> On Thu, Apr 16, 2009 at 1:09 AM, Prabhuram Mohan
>  wrote:
> 
> 
> Hello,
> 
> I am trying to find a way to this through
> fs_cli
> 1) call out to ClientA (1...@192.168.1.108),
> ClientB (1...@192.168.1.108) & ClientC
> (1...@192.168.1.108)
> 2) Bridge all the 3 legs together into one
> call
> 
> Thanks
> Prabhu
> 
> 
> 
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> 
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Re: [Freeswitch-users] Call bridge in free switch

[Brian West]

First off remove the () around the sofia URI.

/b

On Apr 16, 2009, at 10:33 PM, Prabhuram Mohan wrote:


Hi Mike,

I tried the following command per ur advice.. but getting the error  
CHAN_NOT_IMPLEMENTED


originate (sofia/profile/1...@192.168.1.108) &  
conference(3085-192.168.1.102);


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Call bridge in free switch

[Prabhuram Mohan]

Hi Mike,

I tried the following command per ur advice.. but getting the error
CHAN_NOT_IMPLEMENTED

originate (sofia/profile/1...@192.168.1.108) &
conference(3085-192.168.1.102);


freeswi...@internal> originate (sofia/profile/1...@192.168.1.102) &
conference(3085-192.168.1.102);
-ERR CHAN_NOT_IMPLEMENTED

freeswi...@internal> 2009-04-16 20:28:30 [ERR] switch_core_session.c:303
switch_core_session_outgoing_channel() Could not locate channel type (sofia
2009-04-16 20:28:30 [ERR] switch_ivr_originate.c:1486 switch_ivr_originate()
Cannot create outgoing channel of type [(sofia] cause:
[CHAN_NOT_IMPLEMENTED]
2009-04-16 20:28:30 [DEBUG] switch_ivr_originate.c:2084
switch_ivr_originate() Originate Resulted in Error Cause: 66
[CHAN_NOT_IMPLEMENTED]

Thanks
prabhu

On Thu, Apr 16, 2009 at 4:29 PM, Michael Collins  wrote:

> Do you need to monitor the possible failure of one of these calls? Just
> curious. You can call them individually and drop them into a conference
> right at the FS cmd line:
>
> originate (sofia/profile/1...@192.168.1.108) & conference(myconfname);
> originate (sofia/profile/1...@192.168.1.108) & conference(myconfname);
> originate (sofia/profile/1...@192.168.1.108) & conference(myconfname);
>
> You can control the conference behavior with numerous options. See
> http://wiki.freeswitch.org/wiki/Mod_conference for lots of great
> information.
>
> -MC
>
> On Thu, Apr 16, 2009 at 1:09 AM, Prabhuram Mohan wrote:
>
>> Hello,
>>
>>>
>>> I am trying to find a way to this through fs_cli
>>> 1) call out to ClientA (1...@192.168.1.108), ClientB (1...@192.168.1.108)
>>> & ClientC (1...@192.168.1.108)
>>> 2) Bridge all the 3 legs together into one call
>>>
>>> Thanks
>>> Prabhu
>>>
>>
>>
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Re: [Freeswitch-users] Call bridge in free switch

[Michael Collins]

Do you need to monitor the possible failure of one of these calls? Just
curious. You can call them individually and drop them into a conference
right at the FS cmd line:

originate (sofia/profile/1...@192.168.1.108) & conference(myconfname);
originate (sofia/profile/1...@192.168.1.108) & conference(myconfname);
originate (sofia/profile/1...@192.168.1.108) & conference(myconfname);

You can control the conference behavior with numerous options. See
http://wiki.freeswitch.org/wiki/Mod_conference for lots of great
information.

-MC

On Thu, Apr 16, 2009 at 1:09 AM, Prabhuram Mohan wrote:

> Hello,
>
>>
>> I am trying to find a way to this through fs_cli
>> 1) call out to ClientA (1...@192.168.1.108), ClientB (1...@192.168.1.108)
>> & ClientC (1...@192.168.1.108)
>> 2) Bridge all the 3 legs together into one call
>>
>> Thanks
>> Prabhu
>>
>
>
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Re: [Freeswitch-users] FreeSwitch Complex IVR System

[Michael Collins]

To add to David's comments:

Have a look at ESL, which is the event socket library that the fs devs
created. It's an abstraction layer that makes it easier to use the event
socket with the programming language of your choice. In fact, the program
"fs_cli.c" is a great example of a program that uses the ESL to talk to a
FreeSWITCH server. (fs_cli = FreeSWITCH Command Line Interface program,
kinda like "asterisk -r" if that means anything to you...)

Your project is most definitely possible with FreeSWITCH. A number of people
in the FS community have done bits and pieces of what you've described. Your
big challenge is that you're going to need a person or a team who
understands multiple technology concepts and how to integrate them:
telephony & signaling, socket communications, database management, scripting
and programming, etc.

This is a bold project and I would love to see you use FS to be the
telephony engine. Let us know what you decide. Also, if you need
professional FS assistance you can request it at consult...@freeswitch.org.

-MC

On Thu, Apr 16, 2009 at 8:35 AM, David Knell  wrote:

> Hi Guido,
>
> My preferred way is to talk to FS through its event socket
> interface.  This allows you fully to control FS, whilst giving
> you the power to write the code in whatever language and on
> whatever platform you choose.
>
> The documentation starts here:
> http://wiki.freeswitch.org/wiki/Mod_event_socket
>
> Cheers --
>
> Dave
>
> > Hi @all
> >
> > I have a question about a project I want to realize with FreeSwitch. I
> > want to do a complex IVR System which takes a call, do many things in
> > a MSSQL DB, send some Informations to one or many Middleware Servers
> > via TCP/IP, call one or more mobile phones, the first is able to take
> > the call, it can be that he must be able to hear a prompt before he is
> > actually connected to the first caller, then the conversation must be
> > recorded automatically and during the conversation it must be possible
> > for the called party to redirect the call by dtmf. I know that this is
> > all possible, but I want to know which way is the best to do all this?
> >
> >
> > ___
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> > Freeswitch-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
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Re: [Freeswitch-users] [Remote SIP client] Couple of questions

[Michael Collins]

definitely stop by IRC and talk real-time with others who've dealt with this
kind of thing.
-MC

On Thu, Apr 16, 2009 at 7:44 AM, Fred-145  wrote:

>
>
> mercutioviz wrote:
> > Just following up... did you get these questions ironed out?
>
> Not yet, but I do need to have a clear understanding about how to set
> things
> up when NAT is involved, especially when remote SIP users are also behind a
> NAT router, and especially if they can't make any change to it (eg. staying
> in a hotel, or don't have the skills required to open up ports).
>
> Thank you.
> --
> View this message in context:
> http://www.nabble.com/-Remote-SIP-client--Couple-of-questions-tp22698296p23079426.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Optimum sound file format

[Nik Middleton]

Thanks for this.  One of the servers is using sata and the other scsii
drives, so that may be the problem, I'll give it a go.  Problem seems to
escalate past 200 active calls.  Below that all is well.

 

That said, it could also be a db issue, so I've changed my log tables to
innodb (I'm hoping that now I have row level locking as opposed to table
level it will help)

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 16 April 2009 23:25
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Optimum sound file format

 

Looking at your post, You are already using the best format.
If you do not have a fast filesystem try making a ram disk and play the
files from there instead.

if you *really* want you can use sox to turn them all into raw alaw
files and rename them with a .PCMA extension
to avoid the g711 transconding but g711 to PCM is pretty trivial. it's
more likely a file i/o distress you see.



On Thu, Apr 16, 2009 at 5:04 PM, Nik Middleton
 wrote:

Hi Guys,

 

I'm looking for the optimum audio format when using streamfile in a lua
script.

 

I've found CPU load increases rapidly with the number of threads playing
a .wav file.  Can anyone tell me the optimum when using g711a?

 

Right now the the .wav files are 

 

Audio format: PCM

Sample rate : 8 kHz

Mono

Sample Size: 16 bit

Bit rate  :128kbps

 

Will it help CPU load if I resample to a bit rate of 64kbps and sample
size of 8 bit?

 

I have read that the sample size needs to be 13-14bit  +1 for alaw/ulaw
though

 

Regards,


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Re: [Freeswitch-users] Skypiax as a windows service

[UV]

Ok, I think I know where's the confusion here. Let me clarify:
1. FS run beautifully as a service - that's why I assumed it should work.
2. Skype client runs as a service very well too.
3. When running FS as a service with Skypiax (hence Skypiax as a service),
Skypiax doesn't seem to find the SkypeAPI.

In the Wiki page
http://wiki.freeswitch.org/wiki/Skypiax#Running_Skypiax_on_Windows_as_a_Serv
ice it's says that Running Skypiax on Windows as a Service is "Not yet
written" therefore I assumed it's a known limitation.

Are you saying it isn't?

Anyway, the farming solution you suggested should solve the problem - I'd
assume.

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni
Maruzzelli
Sent: Friday, April 17, 2009 1:10 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Skypiax as a windows service

EG: in the "farm out" scenario there will be FS talking via TCP to a
"farm client" (on local machine or remote). The "farm client" talks
with Skype client instances running on the same machine the "farm
client" is running on.

On Thu, Apr 16, 2009 at 1:47 PM, UV  wrote:
> Decoupling the Skyiax from FS will solve the problem as I assume it'll use
> TCP/IP (winsock) to interface with FS - therefore, I can run it still on
the
> same machine but two separate sessions.

yes, it uses TCP for this. So you would end up with FS (with Skypiax
module) running on RDP while the Skype client instances are running as
services, on the same machine (or in different machines). FS will talk
to Skype client instances via TCP.
Is this acceptable to you?

Other question: why not running FS as a service too? If you run FS as
a service and Skype clients as services, all things would works? Why
you want to use RDP for? (sorry for the silly questions, I just want
to understand better).

> However, I think getting the Skypiax
> to work as a service will be more beneficial regardless if it's decoupled
or
> not.

What do you mean? I believe that Skypiax (as an FS module) works when
FS is run as service. Your problem seems to me that you cannot run
Skype instances under RDP because they cannot access the sound device.
Is this correct?

gm

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Re: [Freeswitch-users] Skypiax as a windows service

[Anthony Minessale]

are you planning on just signaling on TCP or both audio and signalling
cos realtime audio over TCP kinda stinks.

you may find that just running FS as the farm and calling to it with sip is
more or less the same idea with no work ;)


On Thu, Apr 16, 2009 at 10:09 AM, Giovanni Maruzzelli
wrote:

> EG: in the "farm out" scenario there will be FS talking via TCP to a
> "farm client" (on local machine or remote). The "farm client" talks
> with Skype client instances running on the same machine the "farm
> client" is running on.
>
> On Thu, Apr 16, 2009 at 1:47 PM, UV  wrote:
> > Decoupling the Skyiax from FS will solve the problem as I assume it'll
> use
> > TCP/IP (winsock) to interface with FS - therefore, I can run it still on
> the
> > same machine but two separate sessions.
>
> yes, it uses TCP for this. So you would end up with FS (with Skypiax
> module) running on RDP while the Skype client instances are running as
> services, on the same machine (or in different machines). FS will talk
> to Skype client instances via TCP.
> Is this acceptable to you?
>
> Other question: why not running FS as a service too? If you run FS as
> a service and Skype clients as services, all things would works? Why
> you want to use RDP for? (sorry for the silly questions, I just want
> to understand better).
>
> > However, I think getting the Skypiax
> > to work as a service will be more beneficial regardless if it's decoupled
> or
> > not.
>
> What do you mean? I believe that Skypiax (as an FS module) works when
> FS is run as service. Your problem seems to me that you cannot run
> Skype instances under RDP because they cannot access the sound device.
> Is this correct?
>
> gm
>
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Re: [Freeswitch-users] Optimum sound file format

[Anthony Minessale]

Looking at your post, You are already using the best format.
If you do not have a fast filesystem try making a ram disk and play the
files from there instead.

if you *really* want you can use sox to turn them all into raw alaw files
and rename them with a .PCMA extension
to avoid the g711 transconding but g711 to PCM is pretty trivial. it's more
likely a file i/o distress you see.


On Thu, Apr 16, 2009 at 5:04 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:

>  Hi Guys,
>
>
>
> I’m looking for the optimum audio format when using streamfile in a lua
> script.
>
>
>
> I’ve found CPU load increases rapidly with the number of threads playing a
> .wav file.  Can anyone tell me the optimum when using g711a?
>
>
>
> Right now the the .wav files are
>
>
>
> Audio format: PCM
>
> Sample rate : 8 kHz
>
> Mono
>
> Sample Size: 16 bit
>
> Bit rate  :128kbps
>
>
>
> Will it help CPU load if I resample to a bit rate of 64kbps and sample size
> of 8 bit?
>
>
>
> I have read that the sample size needs to be 13-14bit  +1 for alaw/ulaw
> though
>
>
>
> Regards,
>
> ___
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>


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[Freeswitch-users] Optimum sound file format

[Nik Middleton]

Hi Guys,

 

I'm looking for the optimum audio format when using streamfile in a lua
script.

 

I've found CPU load increases rapidly with the number of threads playing
a .wav file.  Can anyone tell me the optimum when using g711a?

 

Right now the the .wav files are 

 

Audio format: PCM

Sample rate : 8 kHz

Mono

Sample Size: 16 bit

Bit rate  :128kbps

 

Will it help CPU load if I resample to a bit rate of 64kbps and sample
size of 8 bit?

 

I have read that the sample size needs to be 13-14bit  +1 for alaw/ulaw
though

 

Regards,

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Re: [Freeswitch-users] Issues detecting DTMF tones

[Brian West]

What are you doing exactly?  Can you provide us an example.

/b

On Apr 16, 2009, at 3:49 PM, Pete Mueller wrote:

Hey guys.  Has anyone else experienced the inability to detect/ 
receive DTMF tones?  Just yesterday I had about 4-5 hours where One  
of my IVR scripts would not detect 1, 2 or 3, but detected the other  
digits perfectly.  If I removed the sound file that was playing, and  
substituted silence it worked, add the sound file in, and it broke.   
I have a strong feeling that this is not an issue with FS, but with  
an upstream system.  But wanted to know if anyone has seen this  
before, and how they went about identifying the culprit and/or  
fixing it.


Some background:
-  Using FS trunk
-  Both legs of the call were via SIP gateway.
-  Setting loglevel to 9 (console and sofia) showed that the  
RTP packets were not received by FS for 1/2/3 but were received for  
other digits

-  Both legs of calls were to/from ATT cell phones
-   Was using session:setInputCallback() to receive tones,  
did not test with playAndGetDigits()


Thanks for any help.
-pete


Brian West
br...@freeswitch.org

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[Freeswitch-users] Issues detecting DTMF tones

[Pete Mueller]

Hey guys.  Has anyone else experienced the inability to detect/receive DTMF
tones?  Just yesterday I had about 4-5 hours where One of my IVR scripts
would not detect 1, 2 or 3, but detected the other digits perfectly.  If I
removed the sound file that was playing, and substituted silence it worked,
add the sound file in, and it broke.  I have a strong feeling that this is
not an issue with FS, but with an upstream system.  But wanted to know if
anyone has seen this before, and how they went about identifying the culprit
and/or fixing it.

 

Some background:

-  Using FS trunk

-  Both legs of the call were via SIP gateway.

-  Setting loglevel to 9 (console and sofia) showed that the RTP
packets were not received by FS for 1/2/3 but were received for other digits

-  Both legs of calls were to/from ATT cell phones

-   Was using session:setInputCallback() to receive tones, did not
test with playAndGetDigits()

 

Thanks for any help.

-pete

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Re: [Freeswitch-users] RTP errors

[Brian West]

Well a little more detail would be great  :P

/b

On Apr 16, 2009, at 2:46 PM, Nik Middleton wrote:


Hi Guys,

I’m getting a few of these errors below

sofia.c:3247 sofia_handle_sip_i_state() Reinvite RTP Error!

Are these caused by a fax machine?  Or am I barking up the wrong tree?

Regards,
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[Freeswitch-users] RTP errors

[Nik Middleton]

Hi Guys,

 

I'm getting a few of these errors below

 

sofia.c:3247 sofia_handle_sip_i_state() Reinvite RTP Error!

 

Are these caused by a fax machine?  Or am I barking up the wrong tree?

 

Regards,

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Re: [Freeswitch-users] ekiga and freeswitch

[Brian West]

I think this is a bug in Ekiga, We do CELT on 114 and Ekiga does Speex  
on 114 I suspect your client is sending speex frames on 114 instead of  
celt frames.


/b

On Apr 16, 2009, at 1:05 PM, e schmidbauer wrote:


i've posted the freeswitch svn trunk console debug and sip trace to
the pastebin.

On Sun, Apr 12, 2009 at 1:34 PM, Brian West   
wrote:

Collect a full sip trace and FULL console debug.  Put it on our
pastebin... Chances are Ekiga is doing something stupid... it usually
does silly things.  Also are you on SVN trunk?

/b

On Apr 12, 2009, at 12:11 PM, e schmidbauer wrote:


Not sure if this is a bug in the program or just in my setup.
I've tried using the svn version of freeswitch (as of yesterday)  
and i

got the exact same error. Any input would be appreciated. Thanks!



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Re: [Freeswitch-users] ekiga and freeswitch

[e schmidbauer]

i've posted the freeswitch svn trunk console debug and sip trace to
the pastebin.

On Sun, Apr 12, 2009 at 1:34 PM, Brian West  wrote:
> Collect a full sip trace and FULL console debug.  Put it on our
> pastebin... Chances are Ekiga is doing something stupid... it usually
> does silly things.  Also are you on SVN trunk?
>
> /b
>
> On Apr 12, 2009, at 12:11 PM, e schmidbauer wrote:
>
>> Not sure if this is a bug in the program or just in my setup.
>> I've tried using the svn version of freeswitch (as of yesterday) and i
>> got the exact same error. Any input would be appreciated. Thanks!
>
>
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Re: [Freeswitch-users] Re-2: FreeSwitch Complex IVR System

[David Knell]

Hi Guido,

The event socket interface will give you DTMF events for bridged calls -
just tried it and it works fine.  There's one mild snag, which is that
outbound sockets (which are easier for inbound call handling) will only
give you events relating to the specific call leg that's attached to
that socket - i.e. you can use an outbound socket app to bridge that leg
to an outbound call leg just fine, but you won't get events related to
that outbound call.

So what we do is use an outbound socket app for call control and
scripting, and have a separate inbound socket app which listens for call
state changes and DTMF on all call legs, and a database table which
glues the two together. 

Cheers --

Dave


> Hi Dave,
> 
> thanks for the answer. I am playing around with FS and Event Socket Library 
> for .NET. I get pretty much to run with this, but the reason why I came from 
> Asterisk to FS is that I cannot get DTMF in a bridged call. I thought that I 
> get an Event as soon one dtmf digit is recognized. Unfortunately this isn't 
> the case.
> 
> If I use the default config files and map the keys with bind_meta_app the 
> dtmf tones are recognized and the function behind the bound app is executed. 
> Is this maybe a bug.
> 
> I have read about mod_managed and that I should use it, but I haven't found 
> anything about the usage of it.
> 
> Any suggestions would help
> 
> thanks...Guido
> 
>  Original Message 
> Subject: Re: [Freeswitch-users] FreeSwitch Complex IVR System (16-Apr-2009 
> 17:35)
> From:David Knell 
> To:  g...@exram.de
> 
> > Hi Guido,
> > 
> > My preferred way is to talk to FS through its event socket
> > interface.  This allows you fully to control FS, whilst giving
> > you the power to write the code in whatever language and on
> > whatever platform you choose.
> > 
> > The documentation starts here:
> > http://wiki.freeswitch.org/wiki/Mod_event_socket
> > 
> > Cheers --
> > 
> > Dave
> > 
> > > Hi @all
> > >  
> > > I have a question about a project I want to realize with FreeSwitch. I
> > > want to do a complex IVR System which takes a call, do many things in
> > > a MSSQL DB, send some Informations to one or many Middleware Servers
> > > via TCP/IP, call one or more mobile phones, the first is able to take
> > > the call, it can be that he must be able to hear a prompt before he is
> > > actually connected to the first caller, then the conversation must be
> > > recorded automatically and during the conversation it must be possible
> > > for the called party to redirect the call by dtmf. I know that this is
> > > all possible, but I want to know which way is the best to do all this?
> > > 
> > > 
> > > ___
> > > 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
> > 
> > 
> > ___
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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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[Freeswitch-users] Re-2: FreeSwitch Complex IVR System

[Guido Kuth]

Hi Dave,

thanks for the answer. I am playing around with FS and Event Socket Library for 
.NET. I get pretty much to run with this, but the reason why I came from 
Asterisk to FS is that I cannot get DTMF in a bridged call. I thought that I 
get an Event as soon one dtmf digit is recognized. Unfortunately this isn't the 
case.

If I use the default config files and map the keys with bind_meta_app the dtmf 
tones are recognized and the function behind the bound app is executed. Is this 
maybe a bug.

I have read about mod_managed and that I should use it, but I haven't found 
anything about the usage of it.

Any suggestions would help

thanks...Guido

 Original Message 
Subject: Re: [Freeswitch-users] FreeSwitch Complex IVR System (16-Apr-2009 
17:35)
From:David Knell 
To:  g...@exram.de

> Hi Guido,
> 
> My preferred way is to talk to FS through its event socket
> interface.  This allows you fully to control FS, whilst giving
> you the power to write the code in whatever language and on
> whatever platform you choose.
> 
> The documentation starts here:
> http://wiki.freeswitch.org/wiki/Mod_event_socket
> 
> Cheers --
> 
> Dave
> 
> > Hi @all
> >  
> > I have a question about a project I want to realize with FreeSwitch. I
> > want to do a complex IVR System which takes a call, do many things in
> > a MSSQL DB, send some Informations to one or many Middleware Servers
> > via TCP/IP, call one or more mobile phones, the first is able to take
> > the call, it can be that he must be able to hear a prompt before he is
> > actually connected to the first caller, then the conversation must be
> > recorded automatically and during the conversation it must be possible
> > for the called party to redirect the call by dtmf. I know that this is
> > all possible, but I want to know which way is the best to do all this?
> > 
> > 
> > ___
> > 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
> 
> 
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Re: [Freeswitch-users] FreeSwitch Complex IVR System

[David Knell]

Hi Guido,

My preferred way is to talk to FS through its event socket
interface.  This allows you fully to control FS, whilst giving
you the power to write the code in whatever language and on
whatever platform you choose.

The documentation starts here:
http://wiki.freeswitch.org/wiki/Mod_event_socket

Cheers --

Dave

> Hi @all
>  
> I have a question about a project I want to realize with FreeSwitch. I
> want to do a complex IVR System which takes a call, do many things in
> a MSSQL DB, send some Informations to one or many Middleware Servers
> via TCP/IP, call one or more mobile phones, the first is able to take
> the call, it can be that he must be able to hear a prompt before he is
> actually connected to the first caller, then the conversation must be
> recorded automatically and during the conversation it must be possible
> for the called party to redirect the call by dtmf. I know that this is
> all possible, but I want to know which way is the best to do all this?
> 
> 
> ___
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Re: [Freeswitch-users] Skypiax as a windows service

[Giovanni Maruzzelli]

EG: in the "farm out" scenario there will be FS talking via TCP to a
"farm client" (on local machine or remote). The "farm client" talks
with Skype client instances running on the same machine the "farm
client" is running on.

On Thu, Apr 16, 2009 at 1:47 PM, UV  wrote:
> Decoupling the Skyiax from FS will solve the problem as I assume it'll use
> TCP/IP (winsock) to interface with FS - therefore, I can run it still on the
> same machine but two separate sessions.

yes, it uses TCP for this. So you would end up with FS (with Skypiax
module) running on RDP while the Skype client instances are running as
services, on the same machine (or in different machines). FS will talk
to Skype client instances via TCP.
Is this acceptable to you?

Other question: why not running FS as a service too? If you run FS as
a service and Skype clients as services, all things would works? Why
you want to use RDP for? (sorry for the silly questions, I just want
to understand better).

> However, I think getting the Skypiax
> to work as a service will be more beneficial regardless if it's decoupled or
> not.

What do you mean? I believe that Skypiax (as an FS module) works when
FS is run as service. Your problem seems to me that you cannot run
Skype instances under RDP because they cannot access the sound device.
Is this correct?

gm

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Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

[Peter Olsson]

I've added this as jira case http://jira.freeswitch.org/browse/MODSOFIA-4

I wasn't sure if it should be under mod_sofia or sofia-sip.

The report has a full debug log attached.

Regards,

Peter Olsson

Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale
Skickat: den 16 april 2009 14:23
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the 
call?

yes open a jira http://jira.freeswitch.org

*attach* the following (do not paste it inline into the comments and give all 
trace files a .txt extension)

repeat the trace you did earlier with more debugging enabled.
 type these 3 cli commands before you call
 sofia profile internal siptrace on
 sofia loglevel all 9
 console loglevel debug




On Thu, Apr 16, 2009 at 2:13 AM, Peter Olsson 
mailto:peter.ols...@visionutveckling.se>> 
wrote:

Allright, I tried this again now, with revision 13042 - it's the same result as 
before.. Should I file a jira case for this?



If you want any more information, or more traces, please get back to me, and 
I'll try to help out as much as possible.





Peter





Från: 
freeswitch-users-boun...@lists.freeswitch.org
 
[mailto:freeswitch-users-boun...@lists.freeswitch.org]
 För Brian West
Skickat: den 15 april 2009 23:21

Till: 
freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds 
the call?



What port are you hitting?  Make sure you turn sip tracing on external and 
internal just in case you're using either or both.



/b



On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote:



I've built using latest trunk now, but I won't be able to test again until 
tomorrow - I'll get back to you after that.

Just to make the scenario a bit more clear;
The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP 
Enablement Services), this one talks UDP to FreeSWITCH. Could this be something 
that causes the problem? I also tried to dial into the dialplan, answer the 
call, and then try to deflect the call using REFER. This didn't create any SIP 
messages either (and nothing happened with the call), so it seems there might 
be a bigger issue than just BYE.

Peter



Brian West

br...@freeswitch.org



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Re: [Freeswitch-users] [Remote SIP client] Couple of questions

[Fred-145]

mercutioviz wrote:
> Just following up... did you get these questions ironed out?

Not yet, but I do need to have a clear understanding about how to set things
up when NAT is involved, especially when remote SIP users are also behind a
NAT router, and especially if they can't make any change to it (eg. staying
in a hotel, or don't have the skills required to open up ports).

Thank you.
-- 
View this message in context: 
http://www.nabble.com/-Remote-SIP-client--Couple-of-questions-tp22698296p23079426.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause

[Mikael Aleksander Bjerkeland]

I think I know a bit more about the problem now. The MEDIA_TIMEOUT
hangup cause is probably coming from the B leg of the call and thus not
visible when I do info or debug on mod_cdr_csv.

I then tried the following after bridge to get it:



However, since that bridge of the call is already hung up I got the
following in reply:

variable_other_leg_hangup_cause: [-ERR No Such Channel!
]

Is there a way to get it from the B leg of the call - assuming that's
where the hangup cause comes from?


Thanks!



El jue, 16-04-2009 a las 16:07 +0200, Mikael Aleksander Bjerkeland
escribió:
> Thanks. I just tested and got some more data but it didn't contain any
> variable containing MEDIA_TIMEOUT. Perhaps it's not really set anywhere?
> variable_hangup_cause and variable_originate_disposition contain
> NORMAL_CLEARING and SUCCESS respectively. I need a var which contains
> the real reason for the hangup of the bridge, which in this case is
> MEDIA_TIMEOUT as you can see from the logs.
> 
> 
> 
> 
> El jue, 16-04-2009 a las 07:37 -0500, Anthony Minessale escribió:
> > turn on the debug option in mod_cdr_csv and you will get something
> > similar to the info app only at the end of the call
> > 
> > 
> > On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland
> >  wrote:
> > El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland
> > escribió:
> > 
> > > Hi,
> > >
> > > I have two scenarios I'm having trouble figuring out and I'd
> > be happy
> > > if someone could tell me what I'm doing wrong.
> > >
> > > 1. leg_delay_start=N not working
> > >
> > > I am trying to delay the origination of the second leg in a
> > forked
> > > dial with the following:
> > >
> > >  > >
> > 
> > data="user/mikael-no...@voip.domain.com,[leg_delay_start=10]openzap/1/a/99355151"/>
> > >
> > >
> > > However the second leg is called at exactly the same time as
> > the first
> > > one. I am away from my testing environment right now, so I'm
> > sorry for
> > > not posting my logs. It appears to me that leg_delay_start
> > is broken
> > > on at least rev 13013.
> > >
> > >
> > > 2. I'd like to stop processing the dialplan after a bridge,
> > but not on
> > > specific hangup causes. If I get a MEDIA_TIMEOUT hangup
> > cause in the
> > > call I'd like to continue in the dialplan. Currently I have
> > the
> > > following:
> > >
> > >  > data="hangup_after_bridge=true"/>
> > >  > data="continue_on_fail=true"/>
> > >  > > data="user/mikael-no...@voip.domain.com"/>
> > > 
> > >  > data="openzap/1/a/99355151"/>
> > >
> > >
> > > Any ideas on how to accomplish this?
> > 
> > 
> > I started testing this with the following dialplan:
> > 
> >
> >  
> > > data="hangup_after_bridge=false"/>
> > > data="continue_on_fail=true"/>
> > > 
> > data="user/mikael-no...@fs.voip.domain.com"/>
> >
> > > data="followme_extension=99355151"/>
> > > data="post_call_followme_check"/>
> >
> >  
> >
> > 
> >  
> > > expression="^post_call_followme_check$"/>
> > > expression="^MEDIA_TIMEOUT|$${continue_on_fail_causes}$"
> > break="on-true">
> >  
> >   > data="${followme_extension}"/>
> >
> >
> >  
> >  
> >
> >  
> > 
> > 
> > ${originate_disposition} never has the value of MEDIA_TIMEOUT
> > since the
> > call was answered, which is absolutely correct, so what I am
> > searching
> > for now is how to get the actual hangup cause. The info app
> > doesn't show
> > MEDIA_TIMEOUT anywhere, but my logs show this:
> > 
> > 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377
> > audio_bridge_thread()
> > sofia/internal/sip:mikael-no...@10.247.3.253
> > ending bridge by request from read function
> > 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
> > audio_bridge_thread() Send signal
> > sofia/internal/sip:mikael-no...@10.247.3.253 [BREAK]
> > 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452
> > audio_bridge_thread() BRIDGE THREAD DONE
> > [sofia/internal/sip:mikael-no...@10.247.3.253]
> > 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
> > audio_bridge_thread() Send signal
>

Re: [Freeswitch-users] leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause

[Mikael Aleksander Bjerkeland]

Thanks. I just tested and got some more data but it didn't contain any
variable containing MEDIA_TIMEOUT. Perhaps it's not really set anywhere?
variable_hangup_cause and variable_originate_disposition contain
NORMAL_CLEARING and SUCCESS respectively. I need a var which contains
the real reason for the hangup of the bridge, which in this case is
MEDIA_TIMEOUT as you can see from the logs.




El jue, 16-04-2009 a las 07:37 -0500, Anthony Minessale escribió:
> turn on the debug option in mod_cdr_csv and you will get something
> similar to the info app only at the end of the call
> 
> 
> On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland
>  wrote:
> El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland
> escribió:
> 
> > Hi,
> >
> > I have two scenarios I'm having trouble figuring out and I'd
> be happy
> > if someone could tell me what I'm doing wrong.
> >
> > 1. leg_delay_start=N not working
> >
> > I am trying to delay the origination of the second leg in a
> forked
> > dial with the following:
> >
> >  >
> 
> data="user/mikael-no...@voip.domain.com,[leg_delay_start=10]openzap/1/a/99355151"/>
> >
> >
> > However the second leg is called at exactly the same time as
> the first
> > one. I am away from my testing environment right now, so I'm
> sorry for
> > not posting my logs. It appears to me that leg_delay_start
> is broken
> > on at least rev 13013.
> >
> >
> > 2. I'd like to stop processing the dialplan after a bridge,
> but not on
> > specific hangup causes. If I get a MEDIA_TIMEOUT hangup
> cause in the
> > call I'd like to continue in the dialplan. Currently I have
> the
> > following:
> >
> >  data="hangup_after_bridge=true"/>
> >  data="continue_on_fail=true"/>
> >  > data="user/mikael-no...@voip.domain.com"/>
> > 
> >  data="openzap/1/a/99355151"/>
> >
> >
> > Any ideas on how to accomplish this?
> 
> 
> I started testing this with the following dialplan:
> 
>
>  
> data="hangup_after_bridge=false"/>
> data="continue_on_fail=true"/>
> 
> data="user/mikael-no...@fs.voip.domain.com"/>
>
> data="followme_extension=99355151"/>
> data="post_call_followme_check"/>
>
>  
>
> 
>  
> expression="^post_call_followme_check$"/>
> expression="^MEDIA_TIMEOUT|$${continue_on_fail_causes}$"
> break="on-true">
>  
>   data="${followme_extension}"/>
>
>
>  
>  
>
>  
> 
> 
> ${originate_disposition} never has the value of MEDIA_TIMEOUT
> since the
> call was answered, which is absolutely correct, so what I am
> searching
> for now is how to get the actual hangup cause. The info app
> doesn't show
> MEDIA_TIMEOUT anywhere, but my logs show this:
> 
> 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377
> audio_bridge_thread()
> sofia/internal/sip:mikael-no...@10.247.3.253
> ending bridge by request from read function
> 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
> audio_bridge_thread() Send signal
> sofia/internal/sip:mikael-no...@10.247.3.253 [BREAK]
> 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452
> audio_bridge_thread() BRIDGE THREAD DONE
> [sofia/internal/sip:mikael-no...@10.247.3.253]
> 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
> audio_bridge_thread() Send signal
> sofia/internal/mikael-ek...@fs.voip.domain.com [BREAK]
> 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:508
> switch_core_session_run()
> (sofia/internal/sip:mikael-no...@10.247.3.253)
> State EXCHANGE_MEDIA going to sleep
> 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:397
> switch_core_session_run()
> (sofia/internal/sip:mikael-no...@10.247.3.253)
> Running State Change CS_HANGUP
> EXECUTE sofia/internal/mikael-ek...@fs.voip.domain.com info()
> 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
> switch_core_session_run()
> (sofia/internal/sip:mikael-no...@10.247.3.253)
> State HANGUP
> 2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup()
> Channel
> so

Re: [Freeswitch-users] Call bridge in free switch

[Szymon Olko]

Prabhuram Mohan pisze:
> Hello,
> 
> 
> I am trying to find a way to this through fs_cli
> 1) call out to ClientA (1...@192.168.1.108
> ), ClientB (1...@192.168.1.108
> ) & ClientC (1...@192.168.1.108
> )
> 2) Bridge all the 3 legs together into one call
> 
> Thanks
> Prabhu

The only way I know to do this is to use conference module.

Then for example:

conference test dial clientA
conference test dial clientB
conference test dial clientC

> 
> 
> 
> 
> 
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Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?

[Will Boyce]

It may be worth looking at http://www.mediawiki.org/wiki/Extension:Pdf_Export 
or http://www.mediawiki.org/wiki/Extension:Pdf_Book 

-- 
Regards, 

Will Boyce  
tel: 07933 515 987 
url: http://willboyce.com 

- "Michael Collins"  wrote: 
| From: "Michael Collins"  
| To: freeswitch-users@lists.freeswitch.org 
| Sent: Wednesday, 15 April, 2009 18:12:26 GMT +00:00 GMT Britain, Ireland, 
Portugal 
| Subject: Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ? 
| 
| Unfortunately this isn't being maintained and Bret didn't give his script to 
any of us. If anyone out there is familiar with converting wiki pages to PDF 
and is willing to pick this up then by all means contact me off list and we'll 
discuss it. 
| 
| -MC 
| 
| 
| On Wed, Apr 15, 2009 at 12:22 AM, Mitul Limbani < mi...@enterux.com > wrote: 
| 

Hello there, 
| 
| In my previous encounter with FreeSwitch, I had found that Bret had posted on 
the Mailing List somewhere about availability of the entire FreeSwitch Wiki 
Documentation on a single PDF, this is useful coz at the offset apart from Wiki 
there is no other offline media to learn it. 
| 
| Is the same PDF available looking at the growth of Wiki pages and the 
updation. 
| 
| I look forward to hear from you guys, 
| 
| Thanks & Regards, 
| Mitul Limbani, 
| Founder & CEO, 
| Enterux Solutions, 
| The Enterprise Linux Company (TM), 
| www.enterux.com 
| +91-9820332422 
| 
| 
Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/ 
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| 
| 
| 
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[Freeswitch-users] Call bridge in free switch

[Prabhuram Mohan]

Hello,

>
> I am trying to find a way to this through fs_cli
> 1) call out to ClientA (1...@192.168.1.108), ClientB (1...@192.168.1.108)
> & ClientC (1...@192.168.1.108)
> 2) Bridge all the 3 legs together into one call
>
> Thanks
> Prabhu
>
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[Freeswitch-users] FreeSwitch Complex IVR System

[Guido Kuth]

Hi @all

I have a question about a project I want to realize with FreeSwitch. I want to 
do a complex IVR System which takes a call, do many things in a MSSQL DB, send 
some Informations to one or many Middleware Servers via TCP/IP, call one or 
more mobile phones, the first is able to take the call, it can be that he must 
be able to hear a prompt before he is actually connected to the first caller, 
then the conversation must be recorded automatically and during the 
conversation it must be possible for the called party to redirect the call by 
dtmf. I know that this is all possible, but I want to know which way is the 
best to do all this?
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Re: [Freeswitch-users] leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause

[Anthony Minessale]

turn on the debug option in mod_cdr_csv and you will get something similar
to the info app only at the end of the call


On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland <
mik...@bjerkeland.com> wrote:

> El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland escribió:
> > Hi,
> >
> > I have two scenarios I'm having trouble figuring out and I'd be happy
> > if someone could tell me what I'm doing wrong.
> >
> > 1. leg_delay_start=N not working
> >
> > I am trying to delay the origination of the second leg in a forked
> > dial with the following:
> >
> >  > data="user/mikael-no...@voip.domain.com
> ,[leg_delay_start=10]openzap/1/a/99355151"/>
> >
> >
> > However the second leg is called at exactly the same time as the first
> > one. I am away from my testing environment right now, so I'm sorry for
> > not posting my logs. It appears to me that leg_delay_start is broken
> > on at least rev 13013.
> >
> >
> > 2. I'd like to stop processing the dialplan after a bridge, but not on
> > specific hangup causes. If I get a MEDIA_TIMEOUT hangup cause in the
> > call I'd like to continue in the dialplan. Currently I have the
> > following:
> >
> > 
> > 
> >  > data="user/mikael-no...@voip.domain.com"/>
> > 
> > 
> >
> >
> > Any ideas on how to accomplish this?
>
> I started testing this with the following dialplan:
>
>
>  
>
> 
> data="user/mikael-no...@fs.voip.domain.com"/>
>
>
> data="post_call_followme_check"/>
>
>  
>
>
>  
> expression="^post_call_followme_check$"/>
> expression="^MEDIA_TIMEOUT|$${continue_on_fail_causes}$"
> break="on-true">
>  
>  
>
>
>  
>  
>
>  
>
>
> ${originate_disposition} never has the value of MEDIA_TIMEOUT since the
> call was answered, which is absolutely correct, so what I am searching
> for now is how to get the actual hangup cause. The info app doesn't show
> MEDIA_TIMEOUT anywhere, but my logs show this:
>
> 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377
> audio_bridge_thread() 
> sofia/internal/sip:mikael-no...@10.247.3.253
> ending bridge by request from read function
> 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
> audio_bridge_thread() Send signal
> sofia/internal/sip:mikael-no...@10.247.3.253[BREAK]
> 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452
> audio_bridge_thread() BRIDGE THREAD DONE
> [sofia/internal/sip:mikael-no...@10.247.3.253
> ]
> 2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
> audio_bridge_thread() Send signal
> sofia/internal/mikael-ek...@fs.voip.domain.com [BREAK]
> 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:508
> switch_core_session_run() 
> (sofia/internal/sip:mikael-no...@10.247.3.253
> )
> State EXCHANGE_MEDIA going to sleep
> 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:397
> switch_core_session_run() 
> (sofia/internal/sip:mikael-no...@10.247.3.253
> )
> Running State Change CS_HANGUP
> EXECUTE sofia/internal/mikael-ek...@fs.voip.domain.com info()
> 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
> switch_core_session_run() 
> (sofia/internal/sip:mikael-no...@10.247.3.253
> )
> State HANGUP
> 2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup() Channel
> sofia/internal/sip:mikael-no...@10.247.3.253hanging
>  up, cause:
> MEDIA_TIMEOUT
> 2009-04-16 10:02:34 [DEBUG] mod_sofia.c:370 sofia_on_hangup() Sending
> BYE to 
> sofia/internal/sip:mikael-no...@10.247.3.253
> 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:46
> switch_core_standard_on_hangup()
> sofia/internal/sip:mikael-no...@10.247.3.253Standard
>  HANGUP, cause:
> MEDIA_TIMEOUT
> 2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
> switch_core_session_run() 
> (sofia/internal/sip:mikael-no...@10.247.3.253
> )
> State HANGUP going to sleep
> 2009-04-16 10:02:34 [INFO] mod_dptools.c:946 info_function()
> CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [sofia/internal/mikael-ek...@fs.voip.domain.com]
> Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
> Call-Direction: [inbound]
> Presence-Call-Direction: [inbound]
> Answer-State: [answered]
> Channel-Read-Codec-Name: [G722]
> Channel-Read-Codec-Rate: [16000]
> Channel-Write-Codec-Name: [G722]
> Channel-Write-Codec-Rate: [16000]
> Caller-Username: [mikael-ekiga]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [Mikael Bjerkeland]
> Caller-Caller-ID-Number: [mikael-ekiga]
> Caller-Network-Addr: [10.0.255.251]
> Caller-Destination-Number: [503]
> Caller-Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
> Caller-Source: [mod_sofia]
> Caller-Context: [customers]
> Caller-Channel-Name: [sofia/internal/mikael-ek...@fs.voip.domain.com]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1239868906687578]
> Caller-Channel-Created-Time: [1239868906687578]
> Caller-Channel-Answered-Time: [1239868911327578]
> Caller-Channel-Progress-Time: [1239868907307602]
> Caller-Cha

Re: [Freeswitch-users] How does FS compare with OpenSIPs?

[Anthony Minessale]

We try not to brag about ourselves or *sell* FreeSWITCH to people.

A SIP proxy like openSIPS and a b2bua/media gateway like FreeSWITCH and
meant to be used together.
There is some overlap in SIP functionality but SIP is just one aspect of
FreeSWITCH where SIP is the only
thing OpenSIPS is for.  That is why you find no comparison because it's like
comparing a bus to a car
they both have wheels they both can provide transportation but they serve
different purposes.

On Thu, Apr 16, 2009 at 7:16 AM, Karl Vesterling  wrote:

> H.323 (mod_opal I think)Skype
> Jingle/Jabber (via mod_dingaling)
>
> Text to speach, speach recognition, and far too many to list.
>
>
> Best Regards,
> Karl J. Vesterling
> k...@ken-ton.com
> 202-461-3231 x0
>
> On Apr 16, 2009, at 7:45 AM, Fred-145 wrote:
>
>
>
> Diego Viola wrote:
>
>
> FreeSWITCH is a B2BUA, OpenSIPS is a SIP proxy.
>
>
>
> Thanks Diego. Based on this list features list, what does FS offer that
> OpenSIPs doesn't?
>
> http://www.opensips.org/index.php?n=Resources.Features
>
> I don't know enough about VoIP etc. to be able to tell, but at first sight,
> it seems like OpenSIPs doesn't really handle PBX features, which would be a
> strong point in favor of FreeSwitch.
> --
> View this message in context:
> http://www.nabble.com/How-does-FS-compare-with-OpenSIPs--tp23074733p23076256.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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>
>
>
>
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>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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Re: [Freeswitch-users] how many simultaneous calls support in freeswitch

[Anthony Minessale]

It's not surprising to us.
90% of people who try to load test manage to do it in unnatural conditions
on on inadequate hardware
and end up saying something similar, then they always come back in 2 weeks
with 2000 calls up.


On Thu, Apr 16, 2009 at 3:36 AM, Martin Fiala  wrote:

> Hello
> I've recently tried to test freeswitch with default configurations
> (just added more users and a regexp match in internal.xml SIP
> switching in dialplan) and it performed quite surprisingly slow.. I
> noticed a large disk swapping activity (CPU at registrations of 50
> clients at 100% load!) and I think it's because of all the default
> settings there (like creating voicemail files for every call etc..).
> At least I hope that's it. I will try making it much simpler and see..
> Else there must be some other issue for sure..
> I don't know if freeswitch has support for generating originating
> calls, but there sure is support for "outbound" connections in means
> of connecting to third party providers etc..
> Afaik, there are two modules for cdr provided, check
> http://wiki.freeswitch.org/wiki/Cdr.
> Martin
>
> On Wed, Apr 15, 2009 at 4:06 PM, Brian West  wrote:
> > You have to determine how far it will scale for YOUR needs nobody can
> answer
> > this question.  It all depends on what YOU are doing with it and how
> crazy
> > wild you go with things in your implementation.  ;)
> > /b
> > On Apr 15, 2009, at 8:59 AM, Parveen Kumar Jain wrote:
> >
> > Hi,
> >
> >I need to develop an IVR application which makes an outbound calls and
> > then plays the some audio file for the user. For this I was trying to
> > evaluate Freeswitch under following criteria:
> >
> >  - Does freeswitch have outbound calls support(is there any conf file
> file
> > avilabel where I just can put some series of no. and freeswitch just
> calls
> > those no. sequentially)?
> >  - If yes, how many simultaneous calls are possible on a simple pentium-4
> > using G711U as a codec(1 GB RAM, 2.2 GHz machine) machine?(I had checked
> the
> > switch.conf file and it says that by default it can support upto 1000
> calls
> > , is it true for a small machine also)? In other terms is Freeswitch is
> > scalable if I need to add more users to call from here?
> >   - Does freeswitch have the support of CDR(call data record)  after
> > succesfull calls ?
> >
> > It will be great help if any of you can comment on these question and it
> can
> > save my several hours of testing before I can make my conclusion :)
> >
> > Best Regards,
> > Parveen Jain
> >
> >
> > Brian West
> > br...@freeswitch.org
> > -- Meet us at ClueCon!  http://www.cluecon.com
> >
> >
> >
> >
> >
> > ___
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> > 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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-- 
Anthony Minessale II

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ClueCon http://www.cluecon.com/

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Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

[Anthony Minessale]

yes open a jira http://jira.freeswitch.org

*attach* the following (do not paste it inline into the comments and give
all trace files a .txt extension)

repeat the trace you did earlier with more debugging enabled.
 type these 3 cli commands before you call
 sofia profile internal siptrace on
 sofia loglevel all 9
 console loglevel debug





On Thu, Apr 16, 2009 at 2:13 AM, Peter Olsson <
peter.ols...@visionutveckling.se> wrote:

>  Allright, I tried this again now, with revision 13042 – it’s the same
> result as before.. Should I file a jira case for this?
>
>
>
> If you want any more information, or more traces, please get back to me,
> and I’ll try to help out as much as possible.
>
>
>
>
>
> Peter
>
>
>
>
>
> *Från:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *För *Brian West
> *Skickat:* den 15 april 2009 23:21
> *Till:* freeswitch-users@lists.freeswitch.org
> *Ämne:* Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH
> nds the call?
>
>
>
> What port are you hitting?  Make sure you turn sip tracing on external and
> internal just in case you're using either or both.
>
>
>
> /b
>
>
>
> On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote:
>
>
>
>  I've built using latest trunk now, but I won't be able to test again
> until tomorrow - I'll get back to you after that.
>
> Just to make the scenario a bit more clear;
> The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server
> (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be
> something that causes the problem? I also tried to dial into the dialplan,
> answer the call, and then try to deflect the call using REFER. This didn't
> create any SIP messages either (and nothing happened with the call), so it
> seems there might be a bigger issue than just BYE.
>
> Peter
>
>
>
> Brian West
>
> br...@freeswitch.org
>
>
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
>
>
>
>
> !DSPAM:49e651b332933023977319!
>
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>


-- 
Anthony Minessale II

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

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
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Re: [Freeswitch-users] How does FS compare with OpenSIPs?

[Karl Vesterling]

H.323 (mod_opal I think)
Skype
Jingle/Jabber (via mod_dingaling)

Text to speach, speach recognition, and far too many to list.


Best Regards,
Karl J. Vesterling
k...@ken-ton.com
202-461-3231 x0

On Apr 16, 2009, at 7:45 AM, Fred-145 wrote:




Diego Viola wrote:


FreeSWITCH is a B2BUA, OpenSIPS is a SIP proxy.



Thanks Diego. Based on this list features list, what does FS offer  
that

OpenSIPs doesn't?

http://www.opensips.org/index.php?n=Resources.Features

I don't know enough about VoIP etc. to be able to tell, but at first  
sight,
it seems like OpenSIPs doesn't really handle PBX features, which  
would be a

strong point in favor of FreeSwitch.
--
View this message in context: 
http://www.nabble.com/How-does-FS-compare-with-OpenSIPs--tp23074733p23076256.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] How does FS compare with OpenSIPs?

[Fred-145]

Diego Viola wrote:
> 
> FreeSWITCH is a B2BUA, OpenSIPS is a SIP proxy.
> 

Thanks Diego. Based on this list features list, what does FS offer that
OpenSIPs doesn't?

http://www.opensips.org/index.php?n=Resources.Features

I don't know enough about VoIP etc. to be able to tell, but at first sight,
it seems like OpenSIPs doesn't really handle PBX features, which would be a
strong point in favor of FreeSwitch.
-- 
View this message in context: 
http://www.nabble.com/How-does-FS-compare-with-OpenSIPs--tp23074733p23076256.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] Skypiax as a windows service

[UV]

Hi Giovanni,

We tried every available fake audio driver (i.e. virtual audio cable) but
with no satisfying results.
As for the Skype as a service - that's not a problem. It's working fine -
but only on Session 0 - which makes it inaccessible for the SkypeAPI from
Skypiax.
The problem is not unique to RDP - but most notable when running the Skypiax
via RDP session.
You'll be able to replicate this problem whenever the Skypiax is not running
in the same session / userID of the Skype.
Decoupling the Skyiax from FS will solve the problem as I assume it'll use
TCP/IP (winsock) to interface with FS - therefore, I can run it still on the
same machine but two separate sessions. However, I think getting the Skypiax
to work as a service will be more beneficial regardless if it's decoupled or
not.

Keep up the good work,

Cheers,
UV


-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni
Maruzzelli
Sent: Wednesday, April 15, 2009 12:28 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Skypiax as a windows service

Hi UV,

seems a difficult one this one.

I have no much experience in RDP/terminal server.

If there is no way to have (or fake) audio driver on RDP/terminal
server apps, probably the Skype clients will not works (as you
experienced).

I'm sure, I've read it (:-) ), that Skype clients can be run on a
Windows machine as services, without any user logged in.

That is what I would explore in the future, just adding the How To to
the wiki page.

What you are experiencing seems to be different, seems to be specific
to the RDP/terminal server usage. I'm I understanding you correctly
(that this is specific to RDP)?

Can you send me more info/hints?

In parallel, I'm slowly working on a way to farm out the Skype clients
from the FS servers, so to have the Skype clients running on different
machines on the same LAN. I've a proof of concept working on Linux for
one channel.

You think this would solve your problems (having the Skype clients
running on separate machines other than the machines running FS)?

I'm just back from Easter vacations, please allow a couple days for
the accumulated backlog ;-)

Thanks a lot for taking the time to explore Skypiax and report this,
gm


Sincerely,

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




On Mon, Apr 13, 2009 at 1:32 PM, UV  wrote:
> Great work on Skypiax, Giovanni.
>
>
>
> We’ve tested it in our lab for sometime and it works very well.
>
> Unfortunately, when we tried deploying it on a production environment
> (running Win2K3 server farm), we ran into a barrier:
>
> FS is running as terminal server console application (to be easily
> maintained remotely by RDP)
> This is because Win2K does not allow RDP to access system console (session
> /userid 0)
> Skype does not work on terminal server due to a well known disappearing
> audio drivers problem, therefore it has to run either as a console or a
> service (both on session 0).
> FS can run well as a windows service
> Skypiax seem to load as service, but it can’t find the skype client and
exit
> with the following error:
>
> 2009-04-13 20:54:14 [ERR] mod_skypiax.c:990 load_config() rev
> 13006M[|37 ][ERRORA  990  ][skype_user    ][-1, 0, 0] Failed
to
> connect to a SKYPE API for interface_id=1, no SKYPE client running, please
> (re)start Skype client. Skypiax exiting
>
>
>
> This situation prevents me to run skypiax in production.
>
>
>
> I understand from the wiki page that windows service is not done yet – so
I
> presume this is a predicted outcome.
>
>
>
> Any idea when and if this is planned to be implemented?
>
>
>
> Keep up the good work!
>
>
>
> Cheers,
>
> UV
>
>
>
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Re: [Freeswitch-users] How does FS compare with OpenSIPs?

[Diego Viola]

FreeSWITCH is a B2BUA, OpenSIPS is a SIP proxy.

Diego

On Thu, Apr 16, 2009 at 5:45 AM, Gilles  wrote:

> Hello
>
> There's an excellent article on FS vs. Asterisk, but unless I missed
> it, there's no equivalent to OpenSIPs (www.opensips.org).
>
> At this point, apart from the fact that OpenSIPs is not available for
> Windows, how does FreeSwitch compare with OpenSIPs, what are the
> strengths and weaknesses of each project? I need to know before
> recommending one or the other.
>
> Thank you for any feedback.
>
>
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Re: [Freeswitch-users] conference from a sip provider

[Jason White]

Antony King  wrote:
> so I've put this in dialplan/public.xml:
> 
> 
>   
> 
>   
> 

Wouldn't it be better to put it in dialplan/public/3xxx-conference.xml (or a
similar file name of your choice)? That way, you could leave public.xml
unmodified, and more easily manage the configuration.


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[Freeswitch-users] How does FS compare with OpenSIPs?

[Gilles]

Hello

There's an excellent article on FS vs. Asterisk, but unless I missed 
it, there's no equivalent to OpenSIPs (www.opensips.org).

At this point, apart from the fact that OpenSIPs is not available for 
Windows, how does FreeSwitch compare with OpenSIPs, what are the 
strengths and weaknesses of each project? I need to know before 
recommending one or the other.

Thank you for any feedback.


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Re: [Freeswitch-users] [ANN] Spice Telephony 0.3 - an open source FreeSWITCH/Erlang callcenter

[Giovanni Maruzzelli]

Impressive!

Congratulations

Sincerely,

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




On Thu, Apr 16, 2009 at 5:27 AM, Andrew Thompson  wrote:
> Well, it's been a few months since I mentioned this project last here,
> so here's an update over my last announcement (see
> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010048.html
> )
>
> Things have improved a *lot* since the last time I mentioned it:
>
> * Support for inbound calls (somehow last time that wasn't even
>  finished)
> * Support for outbound calls
> * Support for ringing agents on a softphone or ringing agents on a POTS
>  line
> * A web interface for managing the system (managing agents, skills,
>  queues, etc)
> * A web interface for agents to manage their state (go released, go
>  idle, indicate when they're done with wrapup, etc)
>
> Along with a boatload of minor features and even more bugfixes.
>
> This release is actually something you can play with without knowing
> Erlang, we've even included a boot script to setup and run everything
> for you. So, if you're interested in a distributed, fault tolerant
> callcenter platform built on top of FreeSWITCH I invite you to check out
> http://wiki.opencsm.org/wiki/index.php/Spice_Telephony , the rest of the
> opencsm.org site and especially the other telephony pages on the wiki
> and, if you're interested, please give it a try and let us know what you
> think. Just remember that it's still very much a work in progress
> (although I hope to be able to give a cool demo at ClueCon in August :) ).
>
> Andrew Thompson - opencsm.org
>
>
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Re: [Freeswitch-users] conference from a sip provider

[Antony King]

Aha - hadn't seen that one.

This was in the log after pressing f8:

--
2009-04-16 09:50:58 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 
0XXX->3001 in context public

 
2009-04-16 09:50:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: 
[public_extensions] destination_number(3001) 
=~ /^(10[01][0-9])$/

2009-04-16 09:50:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 
--

so I've put this in dialplan/public.xml:


  

  


Now it accepts the call:

--
2009-04-16 09:54:28 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 
0XXX->3001 in context public

 
2009-04-16 09:54:28 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: 
[public_conferences] 
destination_number(3001) =~ /^(3[0-9][0-9][0-9])$/  

2009-04-16 09:54:28 [DEBUG] switch_core_state_machine.c:100 
switch_core_standard_on_routing() 
(sofia/external/0...@sipgate.co.uk) State Change CS_ROUTING -> 
CS_EXECUTE  
--

Don't get any audio at the moment, but I think that's a separate problem.

Thanks for the tip,

Antony.


On Wednesday 15 April 2009 16:24:45 Brian West wrote:
> Press F8 and try again... With debug cranked up you'll see more details.
>
> On Apr 15, 2009, at 10:10 AM, Antony King wrote:
> > sumably there's some difference between calls coming in via a
> > gateway and localy generated calls; could someone give me some
> > pointers as to how to get it to accept the call ?
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com

-- 

Antony King - 01908 268 901
Systems Consultant
SolutionTrax Technologies - http://www.solutiontrax.com
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Re: [Freeswitch-users] how many simultaneous calls support in freeswitch

[Martin Fiala]

Hello
I've recently tried to test freeswitch with default configurations
(just added more users and a regexp match in internal.xml SIP
switching in dialplan) and it performed quite surprisingly slow.. I
noticed a large disk swapping activity (CPU at registrations of 50
clients at 100% load!) and I think it's because of all the default
settings there (like creating voicemail files for every call etc..).
At least I hope that's it. I will try making it much simpler and see..
Else there must be some other issue for sure..
I don't know if freeswitch has support for generating originating
calls, but there sure is support for "outbound" connections in means
of connecting to third party providers etc..
Afaik, there are two modules for cdr provided, check
http://wiki.freeswitch.org/wiki/Cdr.
Martin

On Wed, Apr 15, 2009 at 4:06 PM, Brian West  wrote:
> You have to determine how far it will scale for YOUR needs nobody can answer
> this question.  It all depends on what YOU are doing with it and how crazy
> wild you go with things in your implementation.  ;)
> /b
> On Apr 15, 2009, at 8:59 AM, Parveen Kumar Jain wrote:
>
> Hi,
>
>    I need to develop an IVR application which makes an outbound calls and
> then plays the some audio file for the user. For this I was trying to
> evaluate Freeswitch under following criteria:
>
>  - Does freeswitch have outbound calls support(is there any conf file file
> avilabel where I just can put some series of no. and freeswitch just calls
> those no. sequentially)?
>  - If yes, how many simultaneous calls are possible on a simple pentium-4
> using G711U as a codec(1 GB RAM, 2.2 GHz machine) machine?(I had checked the
> switch.conf file and it says that by default it can support upto 1000 calls
> , is it true for a small machine also)? In other terms is Freeswitch is
> scalable if I need to add more users to call from here?
>   - Does freeswitch have the support of CDR(call data record)  after
> succesfull calls ?
>
> It will be great help if any of you can comment on these question and it can
> save my several hours of testing before I can make my conclusion :)
>
> Best Regards,
> Parveen Jain
>
>
> Brian West
> br...@freeswitch.org
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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Re: [Freeswitch-users] leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause

[Mikael Aleksander Bjerkeland]

El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland escribió:
> Hi,
> 
> I have two scenarios I'm having trouble figuring out and I'd be happy
> if someone could tell me what I'm doing wrong.
> 
> 1. leg_delay_start=N not working
> 
> I am trying to delay the origination of the second leg in a forked
> dial with the following:
> 
>  data="user/mikael-no...@voip.domain.com,[leg_delay_start=10]openzap/1/a/99355151"/>
> 
> 
> However the second leg is called at exactly the same time as the first
> one. I am away from my testing environment right now, so I'm sorry for
> not posting my logs. It appears to me that leg_delay_start is broken
> on at least rev 13013.
> 
> 
> 2. I'd like to stop processing the dialplan after a bridge, but not on
> specific hangup causes. If I get a MEDIA_TIMEOUT hangup cause in the
> call I'd like to continue in the dialplan. Currently I have the
> following:
> 
> 
> 
>  data="user/mikael-no...@voip.domain.com"/>
> 
> 
> 
> 
> Any ideas on how to accomplish this?

I started testing this with the following dialplan:


  







  


  


  
  


  
  

  


${originate_disposition} never has the value of MEDIA_TIMEOUT since the
call was answered, which is absolutely correct, so what I am searching
for now is how to get the actual hangup cause. The info app doesn't show
MEDIA_TIMEOUT anywhere, but my logs show this:

2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377
audio_bridge_thread() sofia/internal/sip:mikael-no...@10.247.3.253
ending bridge by request from read function
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/sip:mikael-no...@10.247.3.253 [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452
audio_bridge_thread() BRIDGE THREAD DONE
[sofia/internal/sip:mikael-no...@10.247.3.253]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/mikael-ek...@fs.voip.domain.com [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:508
switch_core_session_run() (sofia/internal/sip:mikael-no...@10.247.3.253)
State EXCHANGE_MEDIA going to sleep
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:397
switch_core_session_run() (sofia/internal/sip:mikael-no...@10.247.3.253)
Running State Change CS_HANGUP
EXECUTE sofia/internal/mikael-ek...@fs.voip.domain.com info()
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run() (sofia/internal/sip:mikael-no...@10.247.3.253)
State HANGUP
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup() Channel
sofia/internal/sip:mikael-no...@10.247.3.253 hanging up, cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:370 sofia_on_hangup() Sending
BYE to sofia/internal/sip:mikael-no...@10.247.3.253
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:46
switch_core_standard_on_hangup()
sofia/internal/sip:mikael-no...@10.247.3.253 Standard HANGUP, cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run() (sofia/internal/sip:mikael-no...@10.247.3.253)
State HANGUP going to sleep
2009-04-16 10:02:34 [INFO] mod_dptools.c:946 info_function()
CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/mikael-ek...@fs.voip.domain.com]
Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Caller-Username: [mikael-ekiga]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [Mikael Bjerkeland]
Caller-Caller-ID-Number: [mikael-ekiga]
Caller-Network-Addr: [10.0.255.251]
Caller-Destination-Number: [503]
Caller-Unique-ID: [d505477c-2a5c-11de-9175-4ba93d212d75]
Caller-Source: [mod_sofia]
Caller-Context: [customers]
Caller-Channel-Name: [sofia/internal/mikael-ek...@fs.voip.domain.com]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1239868906687578]
Caller-Channel-Created-Time: [1239868906687578]
Caller-Channel-Answered-Time: [1239868911327578]
Caller-Channel-Progress-Time: [1239868907307602]
Caller-Channel-Progress-Media-Time: [1239868911327578]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
Other-Leg-Username: [mikael-ekiga]
Other-Leg-Dialplan: [XML]
Other-Leg-Caller-ID-Name: [Mikael Bjerkeland]
Other-Leg-Caller-ID-Number: [21651012]
Other-Leg-Network-Addr: [10.247.3.253]
Other-Leg-Destination-Number: [sip:mikael-no...@10.247.3.253]
Other-Leg-Unique-ID: [d50bf8c4-2a5c-11de-9175-4ba93d212d75]
Other-Leg-Source: [mod_sofia]
Other-Leg-Context: [customers]
Other-Leg-Channel-Name: [sofia/internal/sip:mikael-no...@10.247.3.253]
Other-Leg-Screen-Bit: [true

Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

[Peter Olsson]

Allright, I tried this again now, with revision 13042 - it's the same result as 
before.. Should I file a jira case for this?

If you want any more information, or more traces, please get back to me, and 
I'll try to help out as much as possible.


Peter


Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 15 april 2009 23:21
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds 
the call?

What port are you hitting?  Make sure you turn sip tracing on external and 
internal just in case you're using either or both.

/b

On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote:


I've built using latest trunk now, but I won't be able to test again until 
tomorrow - I'll get back to you after that.

Just to make the scenario a bit more clear;
The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP 
Enablement Services), this one talks UDP to FreeSWITCH. Could this be something 
that causes the problem? I also tried to dial into the dialplan, answer the 
call, and then try to deflect the call using REFER. This didn't create any SIP 
messages either (and nothing happened with the call), so it seems there might 
be a bigger issue than just BYE.

Peter

Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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