[Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH

[Diego Viola]

Hi everyone,

I just found this project which seems to be a new VM for running
languages... the parrot project aims to create a virtual machine for Perl 6
and other dynamic languages. You can take a look at it here:

http://www.parrot.org/

It already supports many different languages:

http://www.parrot.org/languages

And it has a Apache module:

http://www.parrot.org/mod_parrot

I wonder that makes it embeddable... would it be possible to create a
mod_parrot for FreeSWITCH?

Please take this only as a feedback and not as anything else.

Diego
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Re: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH

[Diego Viola]

Here mercutioviz gave me some interesting info about Parrot.

20:47 <@mercutioviz> diegoviola: parrot isn't tied specifically to perl 6
but it is an offshoot of the perl 6 effort
20:47 <@mercutioviz> they were smart to break apart the big project into two
separate projects
20:47 <@mercutioviz> parrot is strictly a virtual machine
20:48 <@mercutioviz> perl 6 is strictly a definition of a scripting language
20:48 <@mercutioviz> that lets people port all sorts of stuff to parrot
without tripping over perl 6
20:49 <@mercutioviz> it also creates freakish possibilities, like calling
perl library functions from a python script

Regards,

Diego

On Fri, Jun 12, 2009 at 8:50 PM, Diego Viola  wrote:

> Hi everyone,
>
> I just found this project which seems to be a new VM for running
> languages... the parrot project aims to create a virtual machine for Perl 6
> and other dynamic languages. You can take a look at it here:
>
> http://www.parrot.org/
>
> It already supports many different languages:
>
> http://www.parrot.org/languages
>
> And it has a Apache module:
>
> http://www.parrot.org/mod_parrot
>
> I wonder that makes it embeddable... would it be possible to create a
> mod_parrot for FreeSWITCH?
>
> Please take this only as a feedback and not as anything else.
>
> Diego
>
>
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Re: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH

[Diego Viola]

They seem to have an embedding API here.

http://docs.parrot.org/parrot/latest/html/docs/pdds/draft/pdd10_embedding.pod.html

Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot
support also.

Diego

On Fri, Jun 12, 2009 at 8:53 PM, Diego Viola  wrote:

> Here mercutioviz gave me some interesting info about Parrot.
>
> 20:47 <@mercutioviz> diegoviola: parrot isn't tied specifically to perl 6
> but it is an offshoot of the perl 6 effort
> 20:47 <@mercutioviz> they were smart to break apart the big project into
> two separate projects
> 20:47 <@mercutioviz> parrot is strictly a virtual machine
> 20:48 <@mercutioviz> perl 6 is strictly a definition of a scripting
> language
> 20:48 <@mercutioviz> that lets people port all sorts of stuff to parrot
> without tripping over perl 6
> 20:49 <@mercutioviz> it also creates freakish possibilities, like calling
> perl library functions from a python script
>
> Regards,
>
> Diego
>
>
> On Fri, Jun 12, 2009 at 8:50 PM, Diego Viola wrote:
>
>> Hi everyone,
>>
>> I just found this project which seems to be a new VM for running
>> languages... the parrot project aims to create a virtual machine for Perl 6
>> and other dynamic languages. You can take a look at it here:
>>
>> http://www.parrot.org/
>>
>> It already supports many different languages:
>>
>> http://www.parrot.org/languages
>>
>> And it has a Apache module:
>>
>> http://www.parrot.org/mod_parrot
>>
>> I wonder that makes it embeddable... would it be possible to create a
>> mod_parrot for FreeSWITCH?
>>
>> Please take this only as a feedback and not as anything else.
>>
>> Diego
>>
>>
>
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Re: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH

[Diego Viola]

No, I'm not a C programmer, just offering the idea (feedback).

On Fri, Jun 12, 2009 at 9:15 PM, Jason White  wrote:

> Diego Viola  wrote:
>
> > Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot
> > support also.
>
> Are you offering to write it?
>
>
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Re: [Freeswitch-users] Parrot (mod_parrot) support in FreeSWITCH

[Diego Viola]

Could I add a bounty?

On Fri, Jun 12, 2009 at 9:17 PM, Diego Viola  wrote:

> No, I'm not a C programmer, just offering the idea (feedback).
>
>
> On Fri, Jun 12, 2009 at 9:15 PM, Jason White  wrote:
>
>> Diego Viola  wrote:
>>
>> > Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot
>> > support also.
>>
>> Are you offering to write it?
>>
>>
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Re: [Freeswitch-users] mod_voicemail accounts on-the-fly

[Diego Viola]

And please contribute things back when you do it.

Diego

On Sat, Jun 13, 2009 at 6:42 PM, Brian West  wrote:

> Then you're free to write your own directory hook and plug it directly
> into the database how ever you wish.  Look how XML_CURL uses this
> interface.
>
> /b
>
> On Jun 13, 2009, at 5:28 PM, Adam Wilt wrote:
>
> > Thanks. I would really like mod_voicemail to be database driven,
> > instead of by XML and cURL.
> > I noticed in the documentation that you can provide an ODBC handle
> > in the voicemail.conf.xml file, and so I tried it with MySQL.
> > It created the two tables, voicemail_msgs and voicemail_prefs
> > properly, and voicemail_msgs works the way I expect; when a new
> > voicemail is created it writes a new record in this table.  But I
> > don't understand the purpose of voicemail_prefs;  when I add a
> > record here with a username and password, mod_voicemail ignores it.
> > So I still have to use the config file or xml_curl to set-up the
> > users. It doesn't seem like mod_voicemail reads from or writes to
> > this table.
>
>
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Re: [Freeswitch-users] is there any available gui for freeswitch using cake php?

[Diego Viola]

I'm currently writing a rails app that uses mod_nibblebill for billing, it's
a calling card app.

Diego

On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz  wrote:

>
> Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully
> developed...
> Is there any GUI with billing options?
>
>
> seven-8 wrote:
> >
> > http://www.tcapi.org/index.php?title=Main_Page
> >
> >
> >
> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote:
> >
> >>
> >> is there any available gui for freeswitch using cake php complete
> >> instead of
> >> wikipbx, spice softphone or pfsense?
> >> --
> >> View this message in context:
> >>
> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html
> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>
> >>
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Re: [Freeswitch-users] is there any available gui for freeswitch using cake php?

[Diego Viola]

I'm currently rewriting the entire thing, it was a commercial app
first, but I'm re-writing it in order to make it open source. It's not
ready yet, as soon as I finish it, I will release it to the public.

Diego

On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz wrote:
>
> Can you share me the link of it so i can try... Please
>
> Diego Viola wrote:
>>
>> I'm currently writing a rails app that uses mod_nibblebill for billing,
>> it's
>> a calling card app.
>>
>> Diego
>>
>> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz  wrote:
>>
>>>
>>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet fully
>>> developed...
>>> Is there any GUI with billing options?
>>>
>>>
>>> seven-8 wrote:
>>> >
>>> > http://www.tcapi.org/index.php?title=Main_Page
>>> >
>>> >
>>> >
>>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote:
>>> >
>>> >>
>>> >> is there any available gui for freeswitch using cake php complete
>>> >> instead of
>>> >> wikipbx, spice softphone or pfsense?
>>> >> --
>>> >> View this message in context:
>>> >>
>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html
>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>> >>
>>> >>
>>> >> ___
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>>> >
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>>>
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Re: [Freeswitch-users] is there any available gui for freeswitch using cake php?

[Diego Viola]

Sure, I will let you know when it's done.

On Tue, Jun 16, 2009 at 5:26 AM, Edmar Cruz wrote:
>
> Thanks for that info... Can you send me this project if and only if it is
> already finished on this email darkl...@yahoo.com? Thanks a lot...
>
>
> Diego Viola wrote:
>>
>> I'm currently rewriting the entire thing, it was a commercial app
>> first, but I'm re-writing it in order to make it open source. It's not
>> ready yet, as soon as I finish it, I will release it to the public.
>>
>> Diego
>>
>> On Mon, Jun 15, 2009 at 11:06 PM, Edmar Cruz wrote:
>>>
>>> Can you share me the link of it so i can try... Please
>>>
>>> Diego Viola wrote:
>>>>
>>>> I'm currently writing a rails app that uses mod_nibblebill for billing,
>>>> it's
>>>> a calling card app.
>>>>
>>>> Diego
>>>>
>>>> On Mon, Jun 15, 2009 at 6:21 AM, Edmar Cruz 
>>>> wrote:
>>>>
>>>>>
>>>>> Yup tcapi is a great cake php GUI for freeswitch but it is not yet
>>>>> fully
>>>>> developed...
>>>>> Is there any GUI with billing options?
>>>>>
>>>>>
>>>>> seven-8 wrote:
>>>>> >
>>>>> > http://www.tcapi.org/index.php?title=Main_Page
>>>>> >
>>>>> >
>>>>> >
>>>>> > On Jun 15, 2009, at 5:59 PM, Edmar Cruz wrote:
>>>>> >
>>>>> >>
>>>>> >> is there any available gui for freeswitch using cake php complete
>>>>> >> instead of
>>>>> >> wikipbx, spice softphone or pfsense?
>>>>> >> --
>>>>> >> View this message in context:
>>>>> >>
>>>>> http://www.nabble.com/is-there-any-available-gui-for-freeswitch-using-cake-php--tp24031900p24031900.html
>>>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>>> >>
>>>>> >>
>>>>> >> ___
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>>>>> >
>>>>> >
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>>>>> >
>>>>> >
>>>>>
>>>>> --
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>>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>>>
>>>>>
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[Freeswitch-users] Which GSM gateway to buy?

[Diego Viola]

Hi everyone,

Can you please recommend me some GSM gateway? I'm currently looking
for a good one to buy... anyone have experience PORTech GSM gateways?
Are they good?

I also need it to work with FS, I'm also kinda new with VoIP hardware.

Thanks,

Diego

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Re: [Freeswitch-users] Which GSM gateway to buy?

[Diego Viola]

I need it for gsm termination, I'd like to start with 8 channels, then 16, etc.

Thanks,

Diego

On Tue, Jun 16, 2009 at 5:48 PM,  wrote:
> I did a fair amount of research into GSM gateways about 8 months ago.  I
> should first ask what are you looking to do with the gateway?
>
> -pete
>
>  Original Message 
> Subject: [Freeswitch-users] Which GSM gateway to buy?
> From: Diego Viola 
> Date: Tue, June 16, 2009 2:39 pm
> To: freeswitch-users@lists.freeswitch.org
>
> Hi everyone,
>
> Can you please recommend me some GSM gateway? I'm currently looking
> for a good one to buy... anyone have experience PORTech GSM gateways?
> Are they good?
>
> I also need it to work with FS, I'm also kinda new with VoIP hardware.
>
> Thanks,
>
> Diego
>
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Re: [Freeswitch-users] Which GSM gateway to buy?

[Diego Viola]

We are a start-up company btw.

On Tue, Jun 16, 2009 at 6:09 PM, EdPimentl wrote:
> For those that understand Portuguese
> http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/
> -E
>
>
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Re: [Freeswitch-users] Which GSM gateway to buy?

[Diego Viola]

So we can't afford the top and the latest hardware.

On Tue, Jun 16, 2009 at 6:21 PM, Diego Viola wrote:
> We are a start-up company btw.
>
> On Tue, Jun 16, 2009 at 6:09 PM, EdPimentl wrote:
>> For those that understand Portuguese
>> http://www.metacafe.com/watch/2700394/webinar_khomp_placas_gsm_anal_gica_e_e1/
>> -E
>>
>>
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Re: [Freeswitch-users] javascript session.execute

[Diego Viola]

Applications are the ones in mod_dptools and FSAPI are mod_commands API
right?

On Wed, Jun 17, 2009 at 9:27 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> 1) look in the wiki and try to understand the difference between
> application and FSAPI functions.
> 2) use apiExecute function from js to execute FSAPI functions such as
> uuid_kill
>
>
>
> On Wed, Jun 17, 2009 at 1:44 AM, Baskar  wrote:
>
>> *Hi,
>>
>> I have some questions can any one assist me and answer this query
>>
>> Question 1: Can we able to execute all the api command through JavaScript
>> using session.execute.
>>
>> Question 2: How to kill the session using uuid_kill whether it is
>> possible. **If yes means how we will use uuid_kill in session*
>> *
>> i have tried some thing like this :
>>
>> session.execute("uuid_kill", "session.uuid");
>>
>> session.execute("uuid_kill session.uuid");
>>
>> but there is no output for the session.execute.
>>
>> Whether it is possible to execute uuid_kill in javascript
>>
>> Thanks in advance
>>
>>
>> Warm Regards,
>> N.Baskar
>>
>> *
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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] Which GSM gateway to buy?

[Diego Viola]

Thanks for the suggestions guys, I think I will go with PORTech for now.

@João Mesquita: Let me know when mod_khomp is done, I might consider
getting some khomps in the future when the module is ready.

Regards,

Diego

On Thu, Jun 18, 2009 at 4:26 AM, Jan Kubr wrote:
> There are gateways that allow you to set your own caller ID? I thought it'd
> always use the number of the SIM.
> Jan
>
> On Thu, Jun 18, 2009 at 12:28 AM, jay binks  wrote:
>>
>> Ive used these in the past.
>>
>> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html
>>
>> sound fine, work well...
>> reliable etc etc..
>>
>> things to watch out for...  :
>>
>> *  cant send your own caller ID from them ( in my experience its locked to
>> the sim )
>> *  your provider might block the IMEI number of the GSM terminal, if they
>> dont like what your doing.
>>
>> just some stuff to consider.
>>
>>
>> Jay
>>
>>
>>
>>
>> 2009/6/18 João Mesquita 
>>>
>>> Pricewise, is it worth it?
>>>
>>> jmesquita
>>>
>>> On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr  wrote:
>>>>
>>>> We plan to buy one of these:
>>>>
>>>> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html
>>>> since you can use SMTP/POP3 to manage SMS.
>>>> Jan
>>>> 2009/6/17 João Mesquita 
>>>>>
>>>>> Guys, I was looking at the advantages and disadvantages of having a GSM
>>>>> gateway vs. a GSM board.
>>>>>
>>>>> The conclusions I get are:
>>>>>
>>>>> Board pros
>>>>>
>>>>> 1. Boards are able to get/send SMS without SIP "tricks"
>>>>> 2. You don't have to make a SIP call to check if channel is available
>>>>> and don't rely o SIP messages to get channel status
>>>>> 3. FS will be able to check for signal level on the board and fire
>>>>> events on pre-defined thresholds.
>>>>>
>>>>> Gateway pros
>>>>>
>>>>> 1. I think of is the a GW can be used by more then one server,
>>>>> therefore, can have failover.
>>>>> 2. A GW is more scalable
>>>>>
>>>>> It would be nice if you, that have already used GSM GWs in production,
>>>>> could comment on this.
>>>>>
>>>>> Thanks,
>>>>>
>>>>> jmesquita
>>>>>
>>>>> On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV  wrote:
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> look at www.kuhnt.com. It´s a german page. There you can find
>>>>>> "Kontakt"
>>>>>> where you can ask for special requirements.
>>>>>>
>>>>>> NOx
>>>>>>
>>>>>>
>>>>>>
>>>>>> Diego Viola wrote:
>>>>>> >
>>>>>> > Hi everyone,
>>>>>> >
>>>>>> > Can you please recommend me some GSM gateway? I'm currently looking
>>>>>> > for a good one to buy... anyone have experience PORTech GSM
>>>>>> > gateways?
>>>>>> > Are they good?
>>>>>> >
>>>>>> > I also need it to work with FS, I'm also kinda new with VoIP
>>>>>> > hardware.
>>>>>> >
>>>>>> > Thanks,
>>>>>> >
>>>>>> > Diego
>>>>>> >
>>>>>> > ___
>>>>>> > 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
>>>>>> >
>>>>>> >
>>>>>>
>>>>>> --
>>>>>> View this message in context:
>>>>>> http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html
>>>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>>>>
>>>>>>
>

Re: [Freeswitch-users] doubt of configuration

[Diego Viola]

Sure, but you need to provide more details, what do you want to do exactly?

On Thu, Jun 18, 2009 at 12:58 PM, Mario Guerra Uzae da Silva - <
mariou...@gmail.com> wrote:

> Hi, I am new user of Freeswitch, I am having trouble doing basic
> configurations. Somebody could help me how to configure a simple extension?
>
> Thanks
>
> sorry for my bad english
>
> --
> Mario Uzae
>
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Re: [Freeswitch-users] Open source Java based inbound event socket library available

[Diego Viola]

Because they probably want a stronger copyleft license?

I prefer the GPL because of that reason.

On Sat, Jun 20, 2009 at 8:40 AM, Raffaele P. Guidi <
raffaele.p.gu...@gmail.com> wrote:

> uhm... nice! But why not MPL license (the same as freeswitch)?
>
>
> On Fri, Jun 19, 2009 at 04:06, paul.d...@gmail.com wrote:
>
>> http://versafon.com/versafonweb/Software.jsp
>>
>> Essentially it's a wrapper around inbound socket interface, not all
>> events supported yet, and not all event parameters/variables. It's multi
>> threaded and scaled well in testing.
>> We offer commercial support and development for FreeSwitch as well.
>>
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Re: [Freeswitch-users] If you use mod_nibblebill, please upgrade to rev 13866

[Diego Viola]

Is this issue fixed in latest trunk?

On Sat, Jun 20, 2009 at 3:32 PM, Darren Schreiber  wrote:

>  Hello folks,
> An important stability issue was identified in the FreeSWITCH core
> ODBC drivers when utilizing decimal / float columns in databases (at least
> MySQL anyway). This has an adverse effect on users of mod_nibblebill, as you
> are likely using floating point or decimal columns to track cash amounts.
> FreeSWITCH may periodically core dump when a floating point value is
> retrieved from the database due to this bug.
>
> Please upgrade to rev 13866 (or at least apply the 
> patch
>  -
> it can be applied independently if you are on at least rev 12152). You do
> not need to update mod_nibblebill, as the bug is not actually in
> mod_nibblebill, it just happens to cause the right conditions to exist to
> exhibit this behavior.
>
> Background information is in 
> FSCORE-384
>  .
>
> Thanks to Tony for acting quickly on this patch.
>
> - Darren
>
>
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Re: [Freeswitch-users] If you use mod_nibblebill, please upgrade to rev 13866

[Diego Viola]

Cool, thanks, you guys are fast like the light ;)

On Sat, Jun 20, 2009 at 3:44 PM, Darren Schreiber  wrote:

>  Yes
>
>  --
> *From:* Diego Viola [mailto:diego.vi...@gmail.com]
> *Sent:* Saturday, June 20, 2009 12:41 PM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] If you use mod_nibblebill,please upgrade
> to rev 13866
>
> Is this issue fixed in latest trunk?
>
> On Sat, Jun 20, 2009 at 3:32 PM, Darren Schreiber  wrote:
>
>>  Hello folks,
>> An important stability issue was identified in the FreeSWITCH core
>> ODBC drivers when utilizing decimal / float columns in databases (at least
>> MySQL anyway). This has an adverse effect on users of mod_nibblebill, as you
>> are likely using floating point or decimal columns to track cash amounts.
>> FreeSWITCH may periodically core dump when a floating point value is
>> retrieved from the database due to this bug.
>>
>> Please upgrade to rev 13866 (or at least apply the 
>> patch<http://fisheye.freeswitch.org/browse/FreeSWITCH/src/switch_odbc.c?r1=12152&r2=13866>
>>  -
>> it can be applied independently if you are on at least rev 12152). You do
>> not need to update mod_nibblebill, as the bug is not actually in
>> mod_nibblebill, it just happens to cause the right conditions to exist to
>> exhibit this behavior.
>>
>> Background information is in 
>> FSCORE-384<http://jira.freeswitch.org/browse/FSCORE-384>
>>  .
>>
>> Thanks to Tony for acting quickly on this patch.
>>
>> - Darren
>>
>>
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>
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Re: [Freeswitch-users] Is there any license G729?

[Diego Viola]

Look for this email:

"[Freeswitch-users] Announcement: Howler-optimised G.729A Solution for
FreeSWITCH"

I think the development team is also about to offer G729 licensing if I'm
not mistaken.

On Mon, Jun 29, 2009 at 6:51 AM, Jason White  wrote:

> Edmar Cruz  wrote:
> >
> >   Is there any available license G729 for freeswitch?
>
> Yes. It was announced here a few days ago - see the list archives.
>
>
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Re: [Freeswitch-users] Baby Update!

[Diego Viola]

Congrats!

On Fri, Jul 3, 2009 at 7:35 PM, Brian West  wrote:

> Remember send him a little something to help out with the last minute
> expenses!  ;)
>
> Btw lanboy will be at ClueCon  ;)  As will lanwife!
>
> /b
>
> On Jul 3, 2009, at 6:27 PM, Giovanni Maruzzelli wrote:
>
> > Yeeh!
> >
> >
> > On Fri, Jul 3, 2009 at 10:31 PM, Brian West
> > wrote:
> >> Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz
> >> YAY... Congrats mr Lanman!
> >> /b
> >> On Jul 3, 2009, at 8:58 AM, David Knell wrote:
> >>
> >> On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote:
> >>
> >> Congratulations to Ray and Samantha. Lets see what new features and
> >>
> >> bug fixes we will get in their "new version"..! ;-)
> >>
> >> Bug fixes..?!  I'd refer you to Philip Larkin (went to my school, a
> >> bit
> >> before my time, poet, deceased, recently voted "Britain's favourite
> >> poet") whose "This Be The Verse" suggests otherwise:
> >> http://www.artofeurope.com/larkin/lar2.htm
> >>
> >> [as a recent father myself, I'm trying to prove him wrong..]
> >>
> >> --Dave
> >>
> >> --
> >> David Knell, Director, 3C Limited
> >> T: +44 20 3298 2000
> >> E: d...@3c.co.uk
> >> W: http://www.3c.co.uk
> >>
> >>
> >> ___
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> >>
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[Freeswitch-users] FreeSWITCH article for OSnews

[Diego Viola]

Hey guys.

I wrote the OSnews staff about the possibility to post some FreeSWITCH
articles in the OSnews site, as they have published some Asterisk articles
before, I thought that it would be nice to post something about FreeSWITCH
as well, since it deserves more attention.

This is what they said:

"I don’t think any of the staff have any knowledge on this topic to write an
article, so the best we could do is a page 2 item. As I see it, there’s two
things you can do here:

1. Send the item to us using the "Submit News" link at the top of the page,
it will appear in the back end for all of us to see and one of the staff may
pick it up.

2. Write an article about FreeSWITCH, what it is, and how it differs from
Asterisk using your knowledge an submit it in the usual manner, you’re much
more likely to get on the front page then.

Kind regards,
Kroc."

So if you guys are interested and want to help me to write a FreeSWITCH
article, maybe we could send them so they publish it there.

Diego
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Re: [Freeswitch-users] how to enable ESL for ruby?

[Diego Viola]

Just to mention, there is also a Ruby library here for FreeSWITCH, similar
to ESL, it might interest you.

http://code.rubyists.com/projects/fs
http://github.com/bougyman/freeswitcher/tree/master
http://blog.rubyists.com/2009/05/19/ruby-freeswitch-love

On Wed, Jul 29, 2009 at 12:45 AM, Seven Du  wrote:

> Hi Brian,
>
> Sorry responding late. I still cannot get this work, can you take a look?
>
> http://pastebin.freeswitch.org/9877
>
> Everything works fine on Linux but not on my MAC. I have the default ruby
> framework and port install on /opt/local/bin/ruby, however, even I changed
> the Makefile to use the default ruby framework, it just doesn't work though
> the ESL.so compiled successfully. Where I'm wrong?
>
> Thanks.
>
>
>
> 2009/5/27 Brian West 
>
>>
>> On May 26, 2009, at 11:20 AM, dujinfang wrote:
>>
>> Thanks Brain. Got ESL.so, however on my Mac it is #include 
>> instead of .
>>
>>
>> Actually since we do -framework Ruby it should be ruby/ruby but I think
>> the line above the -framework Ruby should be removed since you're doing i
>> tthe Mac way.
>>
>> /b
>>
>>
>
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Re: [Freeswitch-users] arriving today for ClueCon

[Diego Viola]

Cool, I wish I could be there, next year =D

Any pics or videos of ClueCon? :)

On Mon, Aug 3, 2009 at 8:41 AM, Brian West  wrote:

> Just look for large groups of people with laptops.  I'm sure you can't
> miss us.
>
> /b
>
> On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:
>
> > i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
> > afternoon?  anybody need help with setup?
>
>
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Re: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH Story

[Diego Viola]

Maybe you can link your testimonial or put it here also? :D

http://wiki.freeswitch.org/wiki/Testimonials

On Tue, Aug 4, 2009 at 8:54 PM, Diego Viola  wrote:

> Very cool, and yes, FreeSWITCH does rock =D
>
> Both the software and the community ;)
>
> 2009/8/4 João Mesquita 
>
> If this is good for me to hear, I would imagine to the core team.
>>
>> Despite of this not being a group support meeting, I have to say that:
>> Thank you for sharing, Seven.
>>
>> jmesquita
>>
>> On Tue, Aug 4, 2009 at 4:17 AM, Seven Du  wrote:
>>
>>>  Hello All -   In the spirit of ClueCon (which we are missing this year,
>>> but hopefully not next), we wanted to document our "FreeSWITCH Story".
>>>  We've posted it to the wiki(
>>> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
>>> and it is copied below.
>>>
>>> Thank you all and enjoy a good conference!
>>>
>>> Seven Du (seven)
>>> Jonathan Palley (jpalley_idapted)
>>> Idapted Ltd.
>>>
>>>
>>> *How FreeSWITCH has created hundreds of job opportunities and changed
>>> lives. *
>>>
>>> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
>>> been a key enabler of our business.  We hope this story can be a small way
>>> to say a very big THANK YOU ALL.
>>>
>>> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
>>> really allowed us to do just that.
>>>
>>> What We Do:
>>> We are not a telephony business; we are an educational technology and
>>> service business. In Asia (China, in our case) students must pass English
>>> examinations to study or work abroad and gain new experiences.  However,
>>> there is limited access to native English speakers and the access students
>>> can gain is typically very expensive.  At the same time, in the U.S., there
>>> are many professionals looking for work-at-home opportunities - people who
>>> need jobs and would create great teachers.  Through our technology and
>>> content we empower these people to be effective English teachers.  Does it
>>> work?  Yes.  The majority of our students are getting test scores that many
>>> failed for years to get.  Just hours ago one student called one of our sales
>>> agents crying with joy.  And for our teachers, they are now working in an
>>> industry that was previously unavailable to those living in the U.S.
>>> http://www.idapted.com
>>>
>>> Why FreeSWITCH Enables This:
>>> FreeSWITCH has been a key enabler of our business.  Recording calls,
>>> controlling routing, integrating with various web-based interfaces, enabling
>>> multiple endpoints - these are all key features of what we must do.  Most
>>> importantly, setting up various servers and routes to mitigate cross-Pacific
>>> and country-specific network challenges is key.  Doing what we are doing
>>> with commercial solutions would have made the business unworkable.
>>>
>>> Our Experiences with FreeSWITCH:
>>> We started using FreeSWITCH as our VoIP Platform in April 2008, after
>>> receiving unsatisfactory results with other open source solutions.  It took
>>> one day of reading through the FreeSWITCH source code to know, "this is it.
>>>  This is the VoIP platform we build our business on".  It took a few days of
>>> working with the extremely competent and focused community to re-affirm this
>>> commitment.
>>>
>>> Our Setup:
>>> Our teachers use a custom software that integrates a VoIP client with our
>>> web based platform. Students connect to our teachers "on-demand".  Simply
>>> put, on a web-based comet interface the student enters a phone number (or a
>>> skype name or a gtalk account) and our platform bridges the best available
>>> trainer and the student.  At the same time a web-based interface is being
>>> updated.
>>>
>>> The challenge for us is the connection between teachers and students over
>>> a cross-continent network. For example, we experienced problems earlier this
>>> year when a Asis-Pacific communication fiber broken... So, we've learned to
>>> setup multi servers in multiple datacenters for redundancy.
>>>
>>> We run multi instances of FreeSWITCH so we can always use the cutting
>>> edge and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS)
>>> instance connect to other FreeSWITCHes - Fs-skype

Re: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH Story

[Diego Viola]

Very cool, and yes, FreeSWITCH does rock =D

Both the software and the community ;)

2009/8/4 João Mesquita 

> If this is good for me to hear, I would imagine to the core team.
>
> Despite of this not being a group support meeting, I have to say that:
> Thank you for sharing, Seven.
>
> jmesquita
>
> On Tue, Aug 4, 2009 at 4:17 AM, Seven Du  wrote:
>
>> Hello All -   In the spirit of ClueCon (which we are missing this year,
>> but hopefully not next), we wanted to document our "FreeSWITCH Story".
>>  We've posted it to the wiki(
>> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
>> and it is copied below.
>>
>> Thank you all and enjoy a good conference!
>>
>> Seven Du (seven)
>> Jonathan Palley (jpalley_idapted)
>> Idapted Ltd.
>>
>>
>> *How FreeSWITCH has created hundreds of job opportunities and changed
>> lives. *
>>
>> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
>> been a key enabler of our business.  We hope this story can be a small way
>> to say a very big THANK YOU ALL.
>>
>> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
>> really allowed us to do just that.
>>
>> What We Do:
>> We are not a telephony business; we are an educational technology and
>> service business. In Asia (China, in our case) students must pass English
>> examinations to study or work abroad and gain new experiences.  However,
>> there is limited access to native English speakers and the access students
>> can gain is typically very expensive.  At the same time, in the U.S., there
>> are many professionals looking for work-at-home opportunities - people who
>> need jobs and would create great teachers.  Through our technology and
>> content we empower these people to be effective English teachers.  Does it
>> work?  Yes.  The majority of our students are getting test scores that many
>> failed for years to get.  Just hours ago one student called one of our sales
>> agents crying with joy.  And for our teachers, they are now working in an
>> industry that was previously unavailable to those living in the U.S.
>> http://www.idapted.com
>>
>> Why FreeSWITCH Enables This:
>> FreeSWITCH has been a key enabler of our business.  Recording calls,
>> controlling routing, integrating with various web-based interfaces, enabling
>> multiple endpoints - these are all key features of what we must do.  Most
>> importantly, setting up various servers and routes to mitigate cross-Pacific
>> and country-specific network challenges is key.  Doing what we are doing
>> with commercial solutions would have made the business unworkable.
>>
>> Our Experiences with FreeSWITCH:
>> We started using FreeSWITCH as our VoIP Platform in April 2008, after
>> receiving unsatisfactory results with other open source solutions.  It took
>> one day of reading through the FreeSWITCH source code to know, "this is it.
>>  This is the VoIP platform we build our business on".  It took a few days of
>> working with the extremely competent and focused community to re-affirm this
>> commitment.
>>
>> Our Setup:
>> Our teachers use a custom software that integrates a VoIP client with our
>> web based platform. Students connect to our teachers "on-demand".  Simply
>> put, on a web-based comet interface the student enters a phone number (or a
>> skype name or a gtalk account) and our platform bridges the best available
>> trainer and the student.  At the same time a web-based interface is being
>> updated.
>>
>> The challenge for us is the connection between teachers and students over
>> a cross-continent network. For example, we experienced problems earlier this
>> year when a Asis-Pacific communication fiber broken... So, we've learned to
>> setup multi servers in multiple datacenters for redundancy.
>>
>> We run multi instances of FreeSWITCH so we can always use the cutting edge
>> and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
>> connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
>> only loads mod_dingaling. Here is one beauty of FS: We just had to create
>> different conf dirs (/usr/local/freeswitch, /usr/local/skype,
>> /usr/local/gtalk etc). This allows us to run the same code base over
>> different configurations, and call skype and gtalk accounts just like a
>> normal PSTN gateway (sofia/gateway/pstn/ or sofia/gateway/skype/ or
>> sofia/gateway/gtalk/ ). More important, if one FS (say FS-skype) behaves
>> abnormally or crashes, we can easily change to another FS-skype server (we
>> run other servers located in various places in China and HK for
>> redundancy).
>>
>> FS --|
>>  |---PSTN gateways
>>  |--- FS-skype
>>  |--- FS-gtalk
>>  |--- FS-skype2
>>  |--- more ...
>>
>>
>>
>> COMMUNITY:
>>
>> The community's commitment cannot be undervalued.  The insightful, modular
>> design of FreeSWITCH allows anyone to contribute, whereever their skills
>> lie.  It also allows us to easily make modifications to t

Re: [Freeswitch-users] arriving today for ClueCon

[Diego Viola]

Looking forward to this talk =D

"FreeSWITCH: Learning to Think Fourth Dimensionally"

;)

On Tue, Aug 4, 2009 at 9:53 PM, Gabriel Gunderson  wrote:

> On Tue, Aug 4, 2009 at 6:55 PM, Diego Viola wrote:
> > Cool, I wish I could be there, next year =D
>
> Tomorrow is my wedding anniversary 8/5.  If ClueCon keeps getting
> scheduled on that date, I don't know if I'll ever get to go :(
>
>
> > Any pics or videos of ClueCon? :)
>
> Yes, bring on the vids.
>
> Best,
> Gabe
>
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[Freeswitch-users] Looking for some FreeSWITCH job

[Diego Viola]

Hi,

I'm currently looking for some FS jobs, I really need one, I'm currently
unemployed and looking for some serious FreeSWITCH jobs.

Anyone?

P.S: I also do FS and web development with any language, PHP, Ruby, etc.
anything really.

Thanks,

Diego
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Re: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH Story

[Diego Viola]

Very cool, thanks guys, you make FreeSWITCH even better =D

Your story rocks!

On Wed, Aug 5, 2009 at 12:41 AM, Seven Du wrote:
> And I  added this on the wiki page:
> mod_conference and mod_fifo: We also use FreeSWITCH in our office
> environment as a PBX for call center and customer service connected with
> VoIP and PSTN(openzap) gateways. It is integrated into our CRM system
> naturally and just made sales process, business logic and world wide
> conference much more simpler and easier.
> :)
> 7
>
> 2009/8/5 Jonathan Palley 
>>
>> Diego -
>>   Already done.  See the bottom of the page (we linked to another page
>> because of its length)!
>> :)
>> Jonathan Palley
>> Idapted Ltd.
>
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Re: [Freeswitch-users] freepbx for freeswitch

[Diego Viola]

Yay for FreePBX =D

Does Cudatel uses FreeSWITCH as the engine?

On Wed, Aug 5, 2009 at 5:35 PM, Michael Collins wrote:
> Of course, don't forget this:
> http://www.cudatel.com
>
> For those who want a commercial solution built upon FreeSWITCH: You've got
> it!
> -MC
>
> On Wed, Aug 5, 2009 at 4:24 PM, Nik Middleton
>  wrote:
>>
>> I'd heard rumours that this was going to happen and it's great news and
>> good news for FS as well.  With a user friendly front end, FS is sure to
>> fly.  I have no doubt that this will be the first of many.
>>
>> Regards,
>>
>> -Original Message-
>> From: freeswitch-users-boun...@lists.freeswitch.org
>> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
>> Giovanni Maruzzelli
>> Sent: 05 August 2009 22:00
>> To: freeswitch-users@lists.freeswitch.org;
>> freeswitch-...@lists.freeswitch.org
>> Subject: [Freeswitch-users] freepbx for freeswitch
>>
>> Yay!
>>
>> http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
>> ure
>>
>> Darren Schreiber has made the announcement and is doinng a
>> presentation of FreePBX V3 right now at www.cluecon.com.
>>
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Re: [Freeswitch-users] mod_php needed

[Diego Viola]

Damn, looks like we need to learn some formatting =D.

I made some changes on that article to fix the formatting a bit, lets take
care of the wiki as we care about the software :).

On Sat, Aug 8, 2009 at 2:34 PM, William Suffill
wrote:

> http://wiki.freeswitch.org/wiki/PHP_ESL
>
> Formatting could use a bit of help just exported my working document
> to MediaWiki. Hope to add some more information to ESL as I continue
> to do more with it.
>
> I'm going to start a page on fs_ivrd as well. Feedback welcome.
>
> -- W
>
> IRC: wsuff on freenode
>
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Re: [Freeswitch-users] files.freeswitch.org resets connection.

[Diego Viola]

I been having this problem also.

On Mon, Aug 10, 2009 at 1:26 AM, Godson Gera  wrote:

> Hi FS Team,
>
>
> The files.freeswitch.org is resetting connection since 3 days. As
> a result I was not able to download latest release of FS. Got the trunk
> version from svn. But still it suffers from the lack of sound files.  When
> ever I do 'make uhd-sounds-install' http://files.freeswitch.org resets
> connection immediately wget tries 20 times and gives up. Other users on IRC
> also reported this issue.
>
> --
> Thanks & Regards,
> Godson Gera
>
>
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Re: [Freeswitch-users] files.freeswitch.org resets connection.

[Diego Viola]

http://files-sync.freeswitch.org/

On Mon, Aug 10, 2009 at 1:57 AM, Diego Viola  wrote:

> I been having this problem also.
>
> On Mon, Aug 10, 2009 at 1:26 AM, Godson Gera  wrote:
>
>> Hi FS Team,
>>
>>
>> The files.freeswitch.org is resetting connection since 3 days. As
>> a result I was not able to download latest release of FS. Got the trunk
>> version from svn. But still it suffers from the lack of sound files.  When
>> ever I do 'make uhd-sounds-install' http://files.freeswitch.org resets
>> connection immediately wget tries 20 times and gives up. Other users on IRC
>> also reported this issue.
>>
>> --
>> Thanks & Regards,
>> Godson Gera
>>
>>
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>>
>
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Re: [Freeswitch-users] files.freeswitch.org resets connection.

[Diego Viola]

The address for me is: 69.174.57.101

On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera  wrote:

> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
> name resolution. But the problem is still there even if I use my ISP's  DNS
> (mostly because its resolving to the same to IP address) .
>
> On Mon, Aug 10, 2009 at 7:01 PM, Brian West  wrote:
>
>> As MikeJ pointed out please report the IP address 
>> files.freeswitch.orgresolves to.  We have that on a content delivery network 
>> so that the files
>> are closer to you geographically and you can download them faster but if
>> you're having an issue I'll need the IP so I can report it correctly.
>> Thanks,
>> Brian
>>
>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>
>> Hi FS Team,
>>
>>
>> The files.freeswitch.org is resetting connection since 3 days. As
>> a result I was not able to download latest release of FS. Got the trunk
>> version from svn. But still it suffers from the lack of sound files.  When
>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>> connection immediately wget tries 20 times and gives up. Other users on IRC
>> also reported this issue.
>>
>> --
>> Thanks & Regards,
>> Godson Gera
>>
>>
>>
>> ___
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>>
>>
>
>
> --
> Thanks & Regards,
> Godson Gera
> http://godson.in
>
>
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Re: [Freeswitch-users] answer command

[Diego Viola]

Michael, you're welcome :).

Milena, answer is a mod_dptools command, you can use it from the XML
dialplan or from the event socket outbound. mod_commands API are APIs that
you execute from the socket, event socket inbound, etc. But you can also
execute them from event socket outbound using the "api" command.

I hope that makes sense, correct me if I'm wrong =D.

On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins  wrote:

>
>
> On Tue, Aug 11, 2009 at 9:05 AM, Milena  wrote:
>
>>
>> Hello Brian,
>>
>> I wanna fix the wiki, but to make sure i got it right, does it only work
>> on outbound event socket? or is there any other scenario where it would
>> work.
>>
>
> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
> -MC
>
>
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Re: [Freeswitch-users] answer command

[Diego Viola]

I suggest that you learn the differences between mod_commands commands and
mod_dptools applications, and also the interfaces where you can access and
use them.

As said before, mod_dptools is accessible from dialplan, event socket
outbound, etc. and mod_commands is accessible from the CLI, event socket
(inbound/outbound), XML RPC, etc.

That's all described in the wiki I think.

Let us know if you have any questions =D.

On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola  wrote:

> Michael, you're welcome :).
>
> Milena, answer is a mod_dptools command, you can use it from the XML
> dialplan or from the event socket outbound. mod_commands API are APIs that
> you execute from the socket, event socket inbound, etc. But you can also
> execute them from event socket outbound using the "api" command.
>
> I hope that makes sense, correct me if I'm wrong =D.
>
> On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins wrote:
>
>>
>>
>> On Tue, Aug 11, 2009 at 9:05 AM, Milena  wrote:
>>
>>>
>>> Hello Brian,
>>>
>>> I wanna fix the wiki, but to make sure i got it right, does it only work
>>> on outbound event socket? or is there any other scenario where it would
>>> work.
>>>
>>
>> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
>> -MC
>>
>>
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Re: [Freeswitch-users] files.freeswitch.org resets connection.

[Diego Viola]

It resolves fine for me already.

On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola  wrote:

> The address for me is: 69.174.57.101
>
>
> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera  wrote:
>
>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
>> name resolution. But the problem is still there even if I use my ISP's  DNS
>> (mostly because its resolving to the same to IP address) .
>>
>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West  wrote:
>>
>>> As MikeJ pointed out please report the IP address 
>>> files.freeswitch.orgresolves to.  We have that on a content delivery 
>>> network so that the files
>>> are closer to you geographically and you can download them faster but if
>>> you're having an issue I'll need the IP so I can report it correctly.
>>> Thanks,
>>> Brian
>>>
>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>
>>> Hi FS Team,
>>>
>>>
>>> The files.freeswitch.org is resetting connection since 3 days.
>>> As a result I was not able to download latest release of FS. Got the trunk
>>> version from svn. But still it suffers from the lack of sound files.  When
>>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>>> connection immediately wget tries 20 times and gives up. Other users on IRC
>>> also reported this issue.
>>>
>>> --
>>> Thanks & Regards,
>>> Godson Gera
>>>
>>>
>>>
>>> ___
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>>> FreeSWITCH-users@lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>
>>>
>>
>>
>> --
>> Thanks & Regards,
>> Godson Gera
>> http://godson.in
>>
>>
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Re: [Freeswitch-users] files.freeswitch.org resets connection.

[Diego Viola]

Nope.

On Tue, Aug 11, 2009 at 4:01 PM, Brian West  wrote:

> Does it still give you connection refused?
> /b
>
> On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:
>
> It resolves fine for me already.
>
> On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola wrote:
>
>> The address for me is: 69.174.57.101
>>
>>
>> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera  wrote:
>>
>>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
>>> name resolution. But the problem is still there even if I use my ISP's  DNS
>>> (mostly because its resolving to the same to IP address) .
>>>
>>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West wrote:
>>>
>>>> As MikeJ pointed out please report the IP address 
>>>> files.freeswitch.orgresolves to.  We have that on a content delivery 
>>>> network so that the files
>>>> are closer to you geographically and you can download them faster but if
>>>> you're having an issue I'll need the IP so I can report it correctly.
>>>> Thanks,
>>>> Brian
>>>>
>>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>>
>>>> Hi FS Team,
>>>>
>>>>
>>>> The files.freeswitch.org is resetting connection since 3 days.
>>>> As a result I was not able to download latest release of FS. Got the trunk
>>>> version from svn. But still it suffers from the lack of sound files.  When
>>>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>>>> connection immediately wget tries 20 times and gives up. Other users on IRC
>>>> also reported this issue.
>>>>
>>>> --
>>>> Thanks & Regards,
>>>> Godson Gera
>>>>
>>>>
>>>>
>>>> ___
>>>> 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
>>>>
>>>>
>>>
>>>
>>> --
>>> Thanks & Regards,
>>> Godson Gera
>>> http://godson.in
>>>
>>>
>>> ___
>>> 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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>
>
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Re: [Freeswitch-users] files.freeswitch.org resets connection.

[Diego Viola]

Resolving files.freeswitch.org... failed: Temporary failure in name
resolution.

Again...

On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola  wrote:

> Nope.
>
>
> On Tue, Aug 11, 2009 at 4:01 PM, Brian West  wrote:
>
>> Does it still give you connection refused?
>> /b
>>
>> On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:
>>
>> It resolves fine for me already.
>>
>> On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola wrote:
>>
>>> The address for me is: 69.174.57.101
>>>
>>>
>>> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera  wrote:
>>>
>>>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS
>>>> for name resolution. But the problem is still there even if I use my ISP's
>>>> DNS (mostly because its resolving to the same to IP address) .
>>>>
>>>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West wrote:
>>>>
>>>>> As MikeJ pointed out please report the IP address 
>>>>> files.freeswitch.orgresolves to.  We have that on a content delivery 
>>>>> network so that the files
>>>>> are closer to you geographically and you can download them faster but if
>>>>> you're having an issue I'll need the IP so I can report it correctly.
>>>>> Thanks,
>>>>> Brian
>>>>>
>>>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>>>
>>>>> Hi FS Team,
>>>>>
>>>>>
>>>>> The files.freeswitch.org is resetting connection since 3 days.
>>>>> As a result I was not able to download latest release of FS. Got the trunk
>>>>> version from svn. But still it suffers from the lack of sound files.  When
>>>>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>>>>> connection immediately wget tries 20 times and gives up. Other users on 
>>>>> IRC
>>>>> also reported this issue.
>>>>>
>>>>> --
>>>>> Thanks & Regards,
>>>>> Godson Gera
>>>>>
>>>>>
>>>>>
>>>>> ___
>>>>> 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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Thanks & Regards,
>>>> Godson Gera
>>>> http://godson.in
>>>>
>>>>
>>>> ___
>>>> 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
>>>>
>>>>
>>>
>> ___
>> 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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Re: [Freeswitch-users] files.freeswitch.org resets connection.

[Diego Viola]

http://files-sync.freeswitch.org/ works fine.

On Wed, Aug 12, 2009 at 12:56 AM, Diego Viola  wrote:

> Resolving files.freeswitch.org... failed: Temporary failure in name
> resolution.
>
> Again...
>
>
> On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola wrote:
>
>> Nope.
>>
>>
>> On Tue, Aug 11, 2009 at 4:01 PM, Brian West  wrote:
>>
>>> Does it still give you connection refused?
>>> /b
>>>
>>> On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:
>>>
>>> It resolves fine for me already.
>>>
>>> On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola wrote:
>>>
>>>> The address for me is: 69.174.57.101
>>>>
>>>>
>>>> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera wrote:
>>>>
>>>>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS
>>>>> for name resolution. But the problem is still there even if I use my ISP's
>>>>> DNS (mostly because its resolving to the same to IP address) .
>>>>>
>>>>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West wrote:
>>>>>
>>>>>> As MikeJ pointed out please report the IP address
>>>>>> files.freeswitch.org resolves to.  We have that on a content delivery
>>>>>> network so that the files are closer to you geographically and you can
>>>>>> download them faster but if you're having an issue I'll need the IP so I 
>>>>>> can
>>>>>> report it correctly.
>>>>>> Thanks,
>>>>>> Brian
>>>>>>
>>>>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>>>>
>>>>>> Hi FS Team,
>>>>>>
>>>>>>
>>>>>> The files.freeswitch.org is resetting connection since 3
>>>>>> days. As a result I was not able to download latest release of FS. Got 
>>>>>> the
>>>>>> trunk version from svn. But still it suffers from the lack of sound 
>>>>>> files.
>>>>>> When ever I do 'make uhd-sounds-install'http://files.freeswitch.org 
>>>>>> resets
>>>>>> connection immediately wget tries 20 times and gives up. Other users on 
>>>>>> IRC
>>>>>> also reported this issue.
>>>>>>
>>>>>> --
>>>>>> Thanks & Regards,
>>>>>> Godson Gera
>>>>>>
>>>>>>
>>>>>>
>>>>>> ___
>>>>>> 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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Thanks & Regards,
>>>>> Godson Gera
>>>>> http://godson.in
>>>>>
>>>>>
>>>>> ___
>>>>> 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
>>>>>
>>>>>
>>>>
>>> ___
>>> 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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Re: [Freeswitch-users] files.freeswitch.org resets connection.

[Diego Viola]

Aww, ok.

Bad luck to me :).

On Wed, Aug 12, 2009 at 1:57 AM, Brian West  wrote:

> I would have to say its YOUR system and not ours.
> /b
>
> On Aug 11, 2009, at 11:56 PM, Diego Viola wrote:
>
> Resolving files.freeswitch.org... failed: Temporary failure in name
> resolution.
>
> Again...
>
> On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola 
> wrote:
>
>> Nope.
>>
>>
>
> ___
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Re: [Freeswitch-users] ClueCon Presentations - Where?

[Diego Viola]

Just seen Anthony presentation, very cool ;)

Everyone, watch it!

http://files.freeswitch.org/cluecon_2009/presentations/Day%2001%20Presentation%2002.Anthony%20Minessale.1500kbps.mp4

=D

On Wed, Aug 12, 2009 at 5:07 PM, Michael Collins  wrote:

>
>
> On Wed, Aug 12, 2009 at 2:25 PM, Brian West  wrote:
>
>> They are all getting gathered up and put online...
>> files.freeswitch.org/cluecon_2009 just keep an eye there some of the
>> videos are up also.
>> /b
>>
>
> FYI,
>
> I've uploaded the first batch and they should get synched up on
> files.freeswitch.org/cluecon_2009/presentations any time...
> -MC
>
>
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Re: [Freeswitch-users] answer command

[Diego Viola]

Hey Michael,

Just wondering something, I have found that you added
"conference_set_auto_outcall" on the dptools wiki, but I could not find that
function in the mod_dptools.c, shouldn't that be part of the mod_conference
wiki article? =D.

Best regards,

Diego

On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris  wrote:

> Sip does not support this functionality.  The called device would have
> to support this via some other mechanism such as ctsa which I have
> seen recently someone was looking at for freeswitch.  So the first
> issue you must resolve is the called device needs to support some way
> to do this.
>
> Mike
>
> On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov 
> wrote:
>
> >
> > If I have two FS extensions A and B. I'm calling from A to B and
> > want to
> > answer from B-side in my CTI application and to make SIP phone to be
> > synchronised to my CTI application. Is it possible to do it?
> >
> >
> > Brian West-3 wrote:
> >>
> >> Well you can only truly answer an inbound call to FS... you can't
> >> force answer an outbound call.
> >>
> >> /b
> >>
> >> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
> >>
> >>>
> >>> I will try to paraphrase my question.
> >>> Is there any possibility to answer call  from CTI application and
> >>> synchronise answer with answer in SIP client?Maybe we can use SIP
> >>> functions
> >>> in our CTI application instead of FS api commands?
> >>> I'm trying to find the way to make prototype of lineAnswer command
> >>> in TAPI.
> >>
> >>
> >> ___
> >> 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
> >>
> >>
> >
> > --
> > View this message in context:
> http://www.nabble.com/answer-command-tp24912812p24941422.html
> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >
> >
> > ___
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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] answer command

[Diego Viola]

Err, I asked if that was wrong to fix it.

On Thu, Aug 13, 2009 at 2:15 PM, Diego Viola  wrote:

> I was talking with Michael about fixing stuff in the wiki, so I just asked
> to fix that also.
>
>
> On Thu, Aug 13, 2009 at 9:10 AM, Michael Jerris  wrote:
>
>> It probably belongs there.  It's a wiki, feel free to fix it.  What does
>> this have to do with this thread?
>>
>>
>> On Aug 13, 2009, at 4:03 AM, Diego Viola  wrote:
>>
>> Hey Michael,
>>
>> Just wondering something, I have found that you added
>> "conference_set_auto_outcall" on the dptools wiki, but I could not find that
>> function in the mod_dptools.c, shouldn't that be part of the mod_conference
>> wiki article? =D.
>>
>> Best regards,
>>
>> Diego
>>
>> On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris < 
>> m...@jerris.com> wrote:
>>
>>> Sip does not support this functionality.  The called device would have
>>> to support this via some other mechanism such as ctsa which I have
>>> seen recently someone was looking at for freeswitch.  So the first
>>> issue you must resolve is the called device needs to support some way
>>> to do this.
>>>
>>> Mike
>>>
>>> On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov < 
>>> maxim.tsve...@gmail.com>
>>> wrote:
>>>
>>> >
>>> > If I have two FS extensions A and B. I'm calling from A to B and
>>> > want to
>>> > answer from B-side in my CTI application and to make SIP phone to be
>>> > synchronised to my CTI application. Is it possible to do it?
>>> >
>>> >
>>> > Brian West-3 wrote:
>>> >>
>>> >> Well you can only truly answer an inbound call to FS... you can't
>>> >> force answer an outbound call.
>>> >>
>>> >> /b
>>> >>
>>> >> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
>>> >>
>>> >>>
>>> >>> I will try to paraphrase my question.
>>> >>> Is there any possibility to answer call  from CTI application and
>>> >>> synchronise answer with answer in SIP client?Maybe we can use SIP
>>> >>> functions
>>> >>> in our CTI application instead of FS api commands?
>>> >>> I'm trying to find the way to make prototype of lineAnswer command
>>> >>> in TAPI.
>>> >>
>>> >>
>>> >> ___
>>> >> FreeSWITCH-users mailing list
>>> >> 
>>> FreeSWITCH-users@lists.freeswitch.org
>>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> <http://www.freeswitch.org>http://www.freeswitch.org
>>> >>
>>> >>
>>> >
>>> > --
>>> > View this message in context:
>>> <http://www.nabble.com/answer-command-tp24912812p24941422.html>
>>> http://www.nabble.com/answer-command-tp24912812p24941422.html
>>> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>> >
>>> >
>>> > ___
>>> > FreeSWITCH-users mailing list
>>> > 
>>> FreeSWITCH-users@lists.freeswitch.org
>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>> ___
>>> FreeSWITCH-users mailing list
>>>  
>>> FreeSWITCH-users@lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>
>> ___
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>> http://www.freeswitch.org
>>
>>
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>>
>>
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Re: [Freeswitch-users] answer command

[Diego Viola]

I was talking with Michael about fixing stuff in the wiki, so I just asked
to fix that also.

On Thu, Aug 13, 2009 at 9:10 AM, Michael Jerris  wrote:

> It probably belongs there.  It's a wiki, feel free to fix it.  What does
> this have to do with this thread?
>
>
> On Aug 13, 2009, at 4:03 AM, Diego Viola  wrote:
>
> Hey Michael,
>
> Just wondering something, I have found that you added
> "conference_set_auto_outcall" on the dptools wiki, but I could not find that
> function in the mod_dptools.c, shouldn't that be part of the mod_conference
> wiki article? =D.
>
> Best regards,
>
> Diego
>
> On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris < 
> m...@jerris.com> wrote:
>
>> Sip does not support this functionality.  The called device would have
>> to support this via some other mechanism such as ctsa which I have
>> seen recently someone was looking at for freeswitch.  So the first
>> issue you must resolve is the called device needs to support some way
>> to do this.
>>
>> Mike
>>
>> On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov < 
>> maxim.tsve...@gmail.com>
>> wrote:
>>
>> >
>> > If I have two FS extensions A and B. I'm calling from A to B and
>> > want to
>> > answer from B-side in my CTI application and to make SIP phone to be
>> > synchronised to my CTI application. Is it possible to do it?
>> >
>> >
>> > Brian West-3 wrote:
>> >>
>> >> Well you can only truly answer an inbound call to FS... you can't
>> >> force answer an outbound call.
>> >>
>> >> /b
>> >>
>> >> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
>> >>
>> >>>
>> >>> I will try to paraphrase my question.
>> >>> Is there any possibility to answer call  from CTI application and
>> >>> synchronise answer with answer in SIP client?Maybe we can use SIP
>> >>> functions
>> >>> in our CTI application instead of FS api commands?
>> >>> I'm trying to find the way to make prototype of lineAnswer command
>> >>> in TAPI.
>> >>
>> >>
>> >> ___
>> >> FreeSWITCH-users mailing list
>> >> 
>> FreeSWITCH-users@lists.freeswitch.org
>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> <http://www.freeswitch.org>http://www.freeswitch.org
>> >>
>> >>
>> >
>> > --
>> > View this message in context:
>> <http://www.nabble.com/answer-command-tp24912812p24941422.html>
>> http://www.nabble.com/answer-command-tp24912812p24941422.html
>> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >
>> >
>> > ___
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>> > 
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Re: [Freeswitch-users] ClueCon2009 Torrents

[Diego Viola]

Upload the torrent files in http://files.freeswitch.org ;)

On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks  wrote:

> I'd also seed such a torrent.
>
> Please send the link :)
>
>
>
> On 16/08/2009, at 6:34, João Mesquita  wrote:
>
> > I am interested and would also seed to the community
> >
> > On 8/15/09, Gabriel Gunderson  wrote:
> >> On Sat, Aug 15, 2009 at 12:13 PM, Peder
> >> wrote:
> >>> If you want the torrents, email me off list.
> >>
> >> Why off list?  Isn't the point of torrents to have more people
> >> sharing
> >> in the load?
> >>
> >> Gabe
> >>
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> >>
> >
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> >
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Re: [Freeswitch-users] Application Variable list needed

[Diego Viola]

For dialplan apps take a look at.

http://wiki.freeswitch.org/wiki/Mod_dptools

For channel variables:

http://wiki.freeswitch.org/wiki/Channel_Variables

For FS commands:

http://wiki.freeswitch.org/wiki/Mod_commands

You could use transfer in order to send a call to another extension/context.


On Fri, Aug 21, 2009 at 12:54 AM, Ahmed Munir wrote:

> Hi,
> I'm newbie in FreeSwitch, currently I'm replacing asterisk with FreeSwitch.
> I need list of application variables so I can easily translate my asterisk
> configuration in FreeSwitch. Like I don't know how I translate my dial plan
> in FreeSwitch as I like to forward it to another context as sample listed
> below;
>
> exten => _X.,1,Goto(origination_incoming,${EXTEN},1)
> exten => _X.,n,Hangup()
>
> My another request is how can I change extensions.ael in asterisk to
> FreeSwitch? Is there alternative way of doing this? Kindly do let me know.
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
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Re: [Freeswitch-users] Application Variable list needed

[Diego Viola]

Let us know if you have more questions or need more help :)

On Fri, Aug 21, 2009 at 1:35 AM, Diego Viola  wrote:

> For dialplan apps take a look at.
>
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> For channel variables:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables
>
> For FS commands:
>
> http://wiki.freeswitch.org/wiki/Mod_commands
>
> You could use transfer in order to send a call to another
> extension/context.
>
>
> On Fri, Aug 21, 2009 at 12:54 AM, Ahmed Munir wrote:
>
>> Hi,
>> I'm newbie in FreeSwitch, currently I'm replacing asterisk with
>> FreeSwitch. I need list of application variables so I can easily translate
>> my asterisk configuration in FreeSwitch. Like I don't know how I translate
>> my dial plan in FreeSwitch as I like to forward it to another context as
>> sample listed below;
>>
>> exten => _X.,1,Goto(origination_incoming,${EXTEN},1)
>> exten => _X.,n,Hangup()
>>
>> My another request is how can I change extensions.ael in asterisk to
>> FreeSwitch? Is there alternative way of doing this? Kindly do let me know.
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>> ___
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Re: [Freeswitch-users] MFC-R2 support for FreeSWITCH

[Diego Viola]

Nice work, keep up the great work :).

On Fri, Aug 21, 2009 at 6:29 PM, Moises Silva wrote:
> So, I finally took some days to put up OpenR2 working with OpenZAP, which
> means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
> support for. Including Mexico, Brazil, Argentina and others. The stack has
> been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most
> countries that users may be interested in, support for new variants will be
> added on-demand only (in any case users can always tweak the advanced
> configuration file to create their own variants as a last resort).
> I created a wiki page to illustrate the basic
> setup: http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
> Now is time for testing. I just did minimal testing on my development
> environment, no serious testing, and I know that some stuff is not working
> at this point (I had some issues with variable length DNIS and ANI) which
> should be fixed soon.
> If anyone around happens to have an R2 link and wants to test R2 support in
> OpenZAP, I can give them a hand with the configuration and any issues you
> may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
> #openzap as moy.
>
> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
> Canada
> t. 1 905 474 1990 x 128 | e. m...@sangoma.com
>
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Re: [Freeswitch-users] Inboud Call Queue

[Diego Viola]

Hi guys,

I was wondering if some of you run FreeSWITCH on a call center
environment, I ask this because I plan to do that soon and I was
wondering how well mod_fifo works for queues, etc.

Thanks,

Diego

On Thu, May 7, 2009 at 6:08 AM, Saeed Ahmed wrote:
> Thanks Seven I’ll try it very soon.
>
>
>
> 
>
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of seven
> Sent: Thursday, May 07, 2009 5:42 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Inboud Call Queue
>
>
>
> See this:
>
>
>
> http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo
>
>
>
>
>
> On May 6, 2009, at 10:15 PM, Saeed Ahmed wrote:
>
> Thanks Guys
>
>
>
> 
>
> From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] on
> Behalf Of Anthony Minessale
> Sent: Wednesday, May 06, 2009 3:57 PM
> To: freeswitch-us...@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Inboud Call Queue
>
>
>
> I worked on the patch and added it to trunk rev 13240
>
>
> On Wed, May 6, 2009 at 7:53 AM, dujinfang  wrote:
>
> The patch haven't been merged into trunk. It should be as easy as execute
> the following command in the FS source code root dir:
>
>
>
> patch < /tmp/the_patch_file_name.diff
>
>
>
> I will post an example on the wiki when I finished, hope be soon.
>
>
>
> On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote:
>
>
> Hi Seven,
>
> I am exactly looking for this functionality.
>
> Please let me know when you are finished with new queue manager app. I’ll
> try it in my call center.
>
> Regarding Patch: is it already part of SVN trunk? If not then could you help
> me how to install it, I have no programming background.
>
> Many Thanks.
>
>
>
> 
>
> From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] on
> Behalf Of seven
> Sent: Wednesday, May 06, 2009 4:17 AM
> To: freeswitch-us...@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Inboud Call Queue
>
>
>
>
>
> On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote:
>
>
>
> Hi Michael,
>
> Thanks for a quick reply.
>
> I would definitely create a test environment, but my question is that will
> it work in required way?
>
> I read that in Mod_fifo agent has to call in queue but I need that all
> incoming calls are automatically distributed between available agents or if
> all are busy then should go to voicemail.
>
> I'm working on a call center like queue scenario right now, I'm pretty sure
> it call automatically distributed to available agents, but the customer will
> stay in the queue if all agents are busy by default. You can bind a key to
> the channel and play a message repeatedly to guide the customer to voicemail
> by press a key.
>
>
>
> Also maybe you need this patch to make the fifo works as desired.
>
>
>
> http://jira.freeswitch.org/browse/MODAPP-272
>
>
>
> I would join IRC for further assistance.
>
>
>
> Thanks.
>
>
>
> 
>
> From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] on
> Behalf Of Michael Collins
> Sent: Tuesday, May 05, 2009 7:19 PM
> To: freeswitch-us...@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Inboud Call Queue
>
>
>
>
>
> On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed 
> wrote:
>
> Hi All,
>
> In an inbound call center scenario is it possible that customers calls in
> and calls are distributed between online (who are registered on FS and in
> idle state) agents. I saw some on going discussion on list where it looks
> that currently it’s not possible but I am newbie so maybe I didn’t
> understand it well. If it’s possible then please give me a start point that
> how can I implement it.
>
> I would start here:
> http://wiki.freeswitch.org/wiki/Mod_fifo
>
> I strongly recommend that you set up a FreeSWITCH server and play around
> with it so that you can learn the pros and cons of using the FIFO queues. It
> would be best if you could set up a few phones and set them as FIFO agents
> and then have someone help you make test calls so that you can emulate your
> CC environment.
>
> Also, you might want to join us on IRC: #freeswitch on irc.freenode.net -
> there are several users who've had real world experience with mod_fifo and
> they might be in a good position to answer your questions real-time.
>
> -MC
>
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Re: [Freeswitch-users] Inboud Call Queue

[Diego Viola]

Looks nice, is anyone running that in production?

On Sun, Aug 23, 2009 at 3:08 AM, Jason White  wrote:

> Diego Viola  wrote:
> > I was wondering if some of you run FreeSWITCH on a call center
> > environment, I ask this because I plan to do that soon and I was
> > wondering how well mod_fifo works for queues, etc.
>
> This was mentioned on the list once before, and it might be what you want:
> http://wiki.opencsm.org/wiki/index.php/Spice_Telephony
> (Spice Telephony).
>
>
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Re: [Freeswitch-users] memory leak

[Diego Viola]

Cool, feel free to report these bugs in Jira. If they are indeed FreeSWITCH
bugs.

On Fri, Aug 28, 2009 at 12:10 AM, Jason White  wrote:

> Jay Binks  wrote:
>
> > Reason I ask ...   I personally only have a preference for debian,
> > but others may have policy mandated Os's
> > For their companies, and it would be great to have some info about this.
>
> The only problem I've had with FreeSWITCH under Debian Squeeze and Sid
> involves TLS-related segmentation faults that appear to be related to
> something in the version of libssl supplied with Debian. The same problem
> can't be reproduced on Fedora, for example, but it does occur under Debian
> Lenny as well as Debian Squeeze and Sid (i.e., "testing" and "unstable",
> respectively).
>
> besides this, all of the issues that I have encountered turned out to be
> (usually short-lived) bugs in FreeSWITCH or one of the libraries included
> in
> the source tree - they're mostly FreeSWITCH issues.
>
> I should point out that the FreeSWITCH developers are very good at avoiding
> the introduction of bugs into their code and that known bugs get fixed. It
> appears to be an unchanging fact about programming that with a large and
> complex project, even given highly knowledgeable, experienced, committed
> and
> talented developers (as we have with FreeSWITCH), sometimes, bugs do slip
> through.
>
>
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Re: [Freeswitch-users] Hello, and stuff.

[Diego Viola]

Hi Christensen,

Welcome, you made a good choice on FreeSWITCH, and FS is much better at
those things than Asterisk.

Good luck, we are here to help you and tell us your experience later :).

Diego

On Fri, Aug 28, 2009 at 6:14 PM, Christensen Tom wrote:

>  As a background, I ran an asterisk consulting company for about 3 years
> that I gave up on 2 years ago after repeatedly failing to achieve any sort
> of stability on any sort install over about 30 phones, I gave up.
>
> Maybe that was wrong, I am open to the possibility that I just didn't know
> enough and I was building things wrong, but I worked inside the asterisk
> code (which I feel is a hopeless mess), I implemented a few small custom
> features, anyway...
>
> I'm coming back into the VoIP space now, and I'm wondering what sort of
> issues can I expect in trying to pick up and learn freeswitch?  From what
> I've read on the website, it appears to have a much more sane architecture.
> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
> stable and worst general call quality was asterisk.  I constantly contended
> with strange call quality issues in asterisk, lots of echo (even with
> hardware echo cancellation cards), lots of jitter, lots of call break up
> (even on small systems with 10-20 users, using QoS on the network, and in
> general doing everything I could to prioritize voice over anything else).
>
> When I used Cisco call manager and broadsoft, the voice quality issues were
> basically non-existant, as long as the network was running QoS echo,
> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
> does freeswitch show a marked improvement over asterisk in this department?
> As long as you configure QoS and have hardware echo cancellation does it
> actually work reliably?
>
> Thanks for any additional information about freeswitch you can provide as
> well.  I am a software developer primarily by trade, but I do lots of
> consulting type work in the SME space and I've had a couple projects thrown
> to me that require some integration with a phone system, and I just can't in
> good conscience recommend asterisk anymore.
>
>
> -Tom
>
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Re: [Freeswitch-users] Hello, and stuff.

[Diego Viola]

You guys should put your experiences here :).

http://wiki.freeswitch.org/wiki/Testimonials

Diego

On Fri, Aug 28, 2009 at 10:03 PM, Ognjen Seslija  wrote:

> I currently use FS for a managed PBX services for many companies. It's so
> configurable and extensible, I don't think I would change it for anything. I
> have Cisco CME and SCCP phones in my office all interconnected to FS doing
> logic and thinking. I also have Asterisks as PSTN PRI gateways. FS does all
> the job for me in the world, the other two are just used as protocol
> gateways. I wouldn't change FS for anything else now, after a year and the
> half of playing with it. I fully recommend it.
>
> Regards,
> Ognjen
>
> P.S: To be fair, the echo on the hw ec cards is not really Asterisk's
> fault.
>
>
>
> On Fri, Aug 28, 2009 at 8:14 PM, Christensen Tom 
> wrote:
>
>>  As a background, I ran an asterisk consulting company for about 3 years
>> that I gave up on 2 years ago after repeatedly failing to achieve any sort
>> of stability on any sort install over about 30 phones, I gave up.
>>
>> Maybe that was wrong, I am open to the possibility that I just didn't know
>> enough and I was building things wrong, but I worked inside the asterisk
>> code (which I feel is a hopeless mess), I implemented a few small custom
>> features, anyway...
>>
>> I'm coming back into the VoIP space now, and I'm wondering what sort of
>> issues can I expect in trying to pick up and learn freeswitch?  >From what
>> I've read on the website, it appears to have a much more sane architecture.
>> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
>> stable and worst general call quality was asterisk.  I constantly contended
>> with strange call quality issues in asterisk, lots of echo (even with
>> hardware echo cancellation cards), lots of jitter, lots of call break up
>> (even on small systems with 10-20 users, using QoS on the network, and in
>> general doing everything I could to prioritize voice over anything else).
>>
>> When I used Cisco call manager and broadsoft, the voice quality issues
>> were basically non-existant, as long as the network was running QoS echo,
>> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
>> does freeswitch show a marked improvement over asterisk in this department?
>> As long as you configure QoS and have hardware echo cancellation does it
>> actually work reliably?
>>
>> Thanks for any additional information about freeswitch you can provide as
>> well.  I am a software developer primarily by trade, but I do lots of
>> consulting type work in the SME space and I've had a couple projects thrown
>> to me that require some integration with a phone system, and I just can't in
>> good conscience recommend asterisk anymore.
>>
>>
>> -Tom
>>
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Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

[Diego Viola]

Hi David,

What have you used on FS for call center, mod_fifo?

Can you describe your experience with that, I'm currently interested in call
center + FS scenario.

Diego

On Fri, Aug 28, 2009 at 9:01 PM, David Knell  wrote:

> Hi Raimund,
>
> One FreeSWITCH box will be quite enough to handle the call volumes that
> you're talking about, and it ought to be much more stable than the
> Asterisk solution which you've outlined below.
>
> It's probably best to forget about live failover without calls dropping
> - this isn't something that's supported, and there'd be a lot of work to
> do to develop code to keep two boxed in sync.  Once you get used to a
> stable solution - i.e. something which doesn't crash - then live
> failover, HA, etc., will seem somewhat irrelevant.
>
> I recently did some work on an FS-based call center solution - drop me a
> note if I could be of any help with yours.
>
> Cheers --
>
> Dave
>
> > Hello List,
> >
> > I have read the current thread about scalability and I would need some
> > advice about a callcenter setup:
> >
> > First where I come from:
> > I have lot's of problems with an asterisk solution. I have regular
> > crash's and lock-ups, with downgrading and other stuff i got it
> > somewhat stable, but have nevertheless regular hickups. I am desperate
> > and want to get rid of asterisk and I hope that freeSwitch will
> > provide me with a more stable solution.
> >
> >
> > Our Setup (really nothing special):
> > * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3
> > GIG of RAM, Xircom analog switch connected to mobile stations for 4
> > different providers, Digium 4port cards TP400
> > * 8 queues
> > * ~60 agents (which logon, logoff, pause, unpause), not more than 40
> > concurrently online
> > * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
> > * Music on Hold in the call-queues
> > * No special announcement
> > * Transfers between calls in queues and different agents as well as
> > non agents (i mention this because we have transfer related chrashes
> > in asterisk)
> >
> > The current Problems:
> > * Lockups with different causes (ranging from calls not terminated to
> > heavy thread locking through the AMI interface)
> > * Crashes and library aborts (pthread, libc, crashes related to music
> > on hold, app_queue, transfers)
> >
> > We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are
> > now back to 1.4.21.2 (stock debian) as anything beyond that is for
> > whatever reason highly unstable for our szenario. Maybe we should have
> > been segmenting the box into one asterisk dedicted to talking to the
> > hardware, one especial for queue/sip handling, i do not know. (all
> > issues are well documented in issues.asterisk.org, but it seems to be
> > very, very difficult to get to the bottom of them as they exist since
> > 1.4.23 as it seems and are open until know and not fixable since month.)
> >
> >
> > Now, I really would appreciate some success-stories on how you guys
> > managed to get a stable pbx system with freeSWITCH in regard of HA and
> > scalability:
> >
> > * How to segment freeSWITCH? Or is it stable enough to handle all in
> > one for such a szenario as outlined above?
> > * What would be the best strategie for High Availability / Failover?
> >   -> I read in the WIKI (featurelist) that Livemigration of calls
> from
> > one box to another should be possible?
> >   -> I was thinking about using memcached for storing all state
> > information so another freeswitch box can takeover calls from the
> > first box if it dies, is this possible? If so, how?
> >   -> Is there anotherway to somehow configure freeSWITCH that in the
> > event of a crash i do not loose the current established calls?
> >
> > Basically I just want a stable PBX where I do not have to fear every
> > day it will core-dump or abort or Lock up. Is freeSWITCH mature enough
> > so i can sleep at night for at least 3 month without a crash?
> >
> >
> >
> > Thank you for your Time and help in advance, and I am more than
> > willing to take all the information gathered here and create a wiki
> > page to help other people with the same questions/problems.
> >
> > best
> > Ray
> >
> >
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: d...@3c.co.uk
> W: http://www.3c.co.uk
>
>
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Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

[Diego Viola]

Yes, FreeSWITCH is a system that you can trust 100%. I have switched my
Asterisk servers to FreeSWITCH and have peace now.

If I were you I would get rid of Asterisk and use FreeSWITCH, FS will handle
all what you want very well.

And I agree with David, fail-over is kinda irrelevant since the FS doesn't
crash like Asterisk does.

Diego


On Fri, Aug 28, 2009 at 6:52 PM, Raimund Sacherer wrote:

> Hello List,
>
> I have read the current thread about scalability and I would need some
> advice about a callcenter setup:
>
> First where I come from:
> I have lot's of problems with an asterisk solution. I have regular
> crash's and lock-ups, with downgrading and other stuff i got it
> somewhat stable, but have nevertheless regular hickups. I am desperate
> and want to get rid of asterisk and I hope that freeSwitch will
> provide me with a more stable solution.
>
>
> Our Setup (really nothing special):
> * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3
> GIG of RAM, Xircom analog switch connected to mobile stations for 4
> different providers, Digium 4port cards TP400
> * 8 queues
> * ~60 agents (which logon, logoff, pause, unpause), not more than 40
> concurrently online
> * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
> * Music on Hold in the call-queues
> * No special announcement
> * Transfers between calls in queues and different agents as well as
> non agents (i mention this because we have transfer related chrashes
> in asterisk)
>
> The current Problems:
> * Lockups with different causes (ranging from calls not terminated to
> heavy thread locking through the AMI interface)
> * Crashes and library aborts (pthread, libc, crashes related to music
> on hold, app_queue, transfers)
>
> We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are
> now back to 1.4.21.2 (stock debian) as anything beyond that is for
> whatever reason highly unstable for our szenario. Maybe we should have
> been segmenting the box into one asterisk dedicted to talking to the
> hardware, one especial for queue/sip handling, i do not know. (all
> issues are well documented in issues.asterisk.org, but it seems to be
> very, very difficult to get to the bottom of them as they exist since
> 1.4.23 as it seems and are open until know and not fixable since month.)
>
>
> Now, I really would appreciate some success-stories on how you guys
> managed to get a stable pbx system with freeSWITCH in regard of HA and
> scalability:
>
> * How to segment freeSWITCH? Or is it stable enough to handle all in
> one for such a szenario as outlined above?
> * What would be the best strategie for High Availability / Failover?
>-> I read in the WIKI (featurelist) that Livemigration of calls from
> one box to another should be possible?
>-> I was thinking about using memcached for storing all state
> information so another freeswitch box can takeover calls from the
> first box if it dies, is this possible? If so, how?
>-> Is there anotherway to somehow configure freeSWITCH that in the
> event of a crash i do not loose the current established calls?
>
> Basically I just want a stable PBX where I do not have to fear every
> day it will core-dump or abort or Lock up. Is freeSWITCH mature enough
> so i can sleep at night for at least 3 month without a crash?
>
>
>
> Thank you for your Time and help in advance, and I am more than
> willing to take all the information gathered here and create a wiki
> page to help other people with the same questions/problems.
>
> best
> Ray
>
>
> --
> Raimund Sacherer
> -
> RunSolutions
> Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
>
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Re: [Freeswitch-users] Run a command on event

[Diego Viola]

Or you can do something like this with Ruby + FSR:

http://pastebin.freeswitch.org/10184

On Wed, Sep 2, 2009 at 2:27 PM, Mathieu Parent wrote:

> Hi,
>
> I wanted to run an "originate" command when a MESSAGE_WAITING event is
> fired.
>
> Is there a simpler way than creating a daemon listening to the event socket
> ?
>
> Thanks
>
> Mathieu Parent
>
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Re: [Freeswitch-users] Q931 TE State Timer

[Diego Viola]

Great stuff, thanks for your hard work :).

Keep it up.

Regards,

Diego

On Wed, Sep 2, 2009 at 1:44 PM, Helmut Kuper  wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> just to close this thread, I would like to say that after 21000
> pstn-calls and 45 days up time (old stack: max 10 days) the ISDN call
> resource management based on Q931 timers delivered by the *new* Q931
> stack is still clean. Each done call was torn down to the DOWN state
> cleanly.
>
> This doesn't mean that the new stack is done - but for simple calls to
> and from PSTN it works very good compared to the old Q931 stack.
>
> regards
> Helmut
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFKnnak4tZeNddg3dwRAgPEAJsH+wg4CKV2MKQPAUbO69iijR/r4ACgug+a
> zLeZX1bmYNF4UlirU0XJHfc=
> =B/LZ
> -END PGP SIGNATURE-
>
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Re: [Freeswitch-users] E1 Sangoma Card

[Diego Viola]

Good to know, I am getting Sangoma cards for my FreeSWITCH as well.

Diego

On Wed, Sep 2, 2009 at 5:29 PM, Michael Collins  wrote:

>
>
> On Wed, Sep 2, 2009 at 8:52 AM, Eric Richmond  wrote:
>
>> (Sorry if this has been posted before, or even 2x.  I sent it out the
>> first time with the wrong email address, and the 2nd time before i
>> confirmed I wanted to be on the list, so my assumption is that it
>> hasn't gone out yet, although I might be wrong, and if it has gone out
>> before, I'm sorry for spamming the list.  It won't happen in the
>> future.)
>>
>> Hello all,
>>
>> I'm wondering if anyone has successfully deployed a freeswitch server
>> with an E1 card attached.  By looking at the wiki and chatting in IRC,
>> I've heard that theoretically a FS + Sangoma E1 card solution should
>> work, but I'd really like to hear from someone who actually has it
>> working, so I know what gotchas I might encounter.
>>
>> Any insight into this would be great.
>>
>> Thanks,
>>
>> -Eric Richmond
>>
>
> Erich,
>
> Yes, FS + Sangoma E1 card has been used. Is it safe to assume that you need
> it for PRI? Moises from Sangoma has definitely used it and I believe we have
> several people in Europe who are running E1/PRI. Your biggest gotcha is
> probably which PRI stack to use, so for now I would use the libpri method
> that the devs have created. More info here:
> http://wiki.freeswitch.org/wiki/OpenZAP#Adding_libpri_Support
>
> Just curious - what's your application?
> -MC
>
>
>
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Re: [Freeswitch-users] T38 <> T30 transcoding

[Diego Viola]

I believe FreeSWITCH doesn't do T.38 other than pass-through yet, there are
plans for complete support I think but I'm not sure when.

Diego

On Thu, Sep 3, 2009 at 12:29 PM, Tihomir Culjaga  wrote:

> anyone knows anything about this?
>
> T.
>
>
> On Wed, Sep 2, 2009 at 10:53 PM, Tihomir Culjaga wrote:
>
>> I will put several nickels saying it is impossible :)
>>
>>
>> seriously, can it be done?
>>
>> T.
>>
>> On Wed, Sep 2, 2009 at 10:35 PM, Tim Meade wrote:
>>
>>>  I am very interested in a response to this.  Last I knew there was only
>>> T.38 pass through and for some reason I'm not even sure it that was fully
>>> implemented.
>>>
>>>
>>>
>>> Tim
>>>
>>>
>>>
>>> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
>>> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Tihomir
>>> Culjaga
>>> *Sent:* Wednesday, September 02, 2009 3:33 PM
>>> *To:* freeswitch-users@lists.freeswitch.org
>>> *Subject:* [Freeswitch-users] T38 <> T30 transcoding
>>>
>>>
>>>
>>> Hi guys,
>>>
>>> just a quick question... is it possible to do a reliable on the fly T30
>>> <> T38 transcoding at all ... what is the status of T.38 on FS ?
>>>
>>> T,
>>>
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>>
>
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Re: [Freeswitch-users] Presence Feature

[Diego Viola]

Not sure if it does via SIMPLE protocol, but if you do "event plain all" you
should see the PRESENCE_IN and PRESENCE_OUT events.

Diego

On Thu, Sep 3, 2009 at 6:40 PM, Jerry Richards
wrote:

> Does Freeswitch support Presence via SIMPLE protocol?  Can it maintain
> presence?  I presume this would be a SUBSCRIBE/NOTIFY arrangement?
>
> Best Regards,
> Jerry
>
>
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Re: [Freeswitch-users] Presence Feature

[Diego Viola]

I have made a simple script that catches the PRESENCE IN/OUT events and
passes them to a method to get the information for the phones, you can take
a look at it here.

http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/presence/presence.rb?r=14679

Diego

On Thu, Sep 3, 2009 at 7:27 PM, Diego Viola  wrote:

> Not sure if it does via SIMPLE protocol, but if you do "event plain all"
> you should see the PRESENCE_IN and PRESENCE_OUT events.
>
> Diego
>
>
> On Thu, Sep 3, 2009 at 6:40 PM, Jerry Richards  > wrote:
>
>> Does Freeswitch support Presence via SIMPLE protocol?  Can it maintain
>> presence?  I presume this would be a SUBSCRIBE/NOTIFY arrangement?
>>
>> Best Regards,
>> Jerry
>>
>>
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Re: [Freeswitch-users] Digium boards

[Diego Viola]

Hi Pedro,

Yes it should, it does.

Diego

2009/9/3 Pedro Prado 

>  Hi,
>
> Does Freeswith work with any Digium boards?
>
> Ps: Sorry for the newbie question.
>
> Thanks,
> Pedro Prado
>
> --
> Novo Internet Explorer 8: traduza com apenas um clique. Baixe agora, é
> grátis!
>
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Re: [Freeswitch-users] Digium boards

[Diego Viola]

Any Zaptel-compatible card should work.

Diego

On Thu, Sep 3, 2009 at 8:15 PM, Diego Viola  wrote:

> Hi Pedro,
>
> Yes it should, it does.
>
> Diego
>
> 2009/9/3 Pedro Prado 
>
>>  Hi,
>>
>> Does Freeswith work with any Digium boards?
>>
>> Ps: Sorry for the newbie question.
>>
>> Thanks,
>> Pedro Prado
>>
>> --
>> Novo Internet Explorer 8: traduza com apenas um clique. Baixe agora, é
>> grátis!<http://brasil.microsoft.com.br/IE8/mergulhe/?utm_source=MSN%3BHotmail&utm_medium=Tagline&utm_campaign=IE8>
>>
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Re: [Freeswitch-users] Presence Feature

[Diego Viola]

Hi Luis,

My script simply adds two event hooks for presence (PRESENCE_IN and
PRESENCE_OUT) and then it passes those events to a method (presence_handler)
and it prints the "from" which is something like: 1000%192.168.0.2 (the user
and ip) and the status: which can be Registered or Unregistered.

The script is a daemon, so when you register/unregister a phone you see
something like this:

*I, [2009-08-29T03:05:20.351467 #3092]  INFO -- : Phone 1000%192.168.0.2 -
Unregistered

I, [2009-08-29T03:05:20.864802 #3092]  INFO -- : Phone 1000%192.168.0.2 -
Unregistered

I, [2009-08-29T03:05:20.865763 #3092]  INFO -- : Phone 1000%192.168.0.2 -
Registered(UDP)

I, [2009-08-29T03:05:21.390911 #3092]  INFO -- : Phone 1000%192.168.0.2 -
Registered(UDP)

I, [2009-08-29T03:05:21.391308 #3092]  INFO -- : Phone 1000%192.168.0.2 -
Registered(UDP)

I, [2009-08-29T03:05:21.548433 #3092]  INFO -- : Phone 1000%192.168.0.2 -
Unregistered*

That's pretty much what it does, but this makes it possible to make apps to
report the status of phones, etc. I was thinking to send that information on
a database and then show it on a web interface, with Ajax, etc. on "near
real time", it would be pretty cool :-).

Let me know if you have any questions or need any help.

Regards,

Diego

On Thu, Sep 3, 2009 at 8:29 PM, Luis F Urrea  wrote:

> Diego can you explain a little bit further what your script does.
>
> Not really versed into ruby but I certainly interested on these events
>
>
> On Thu, Sep 3, 2009 at 1:30 PM, Diego Viola  wrote:
>
>> I have made a simple script that catches the PRESENCE IN/OUT events and
>> passes them to a method to get the information for the phones, you can take
>> a look at it here.
>>
>>
>> http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/presence/presence.rb?r=14679
>>
>> Diego
>>
>>
>> On Thu, Sep 3, 2009 at 7:27 PM, Diego Viola wrote:
>>
>>> Not sure if it does via SIMPLE protocol, but if you do "event plain all"
>>> you should see the PRESENCE_IN and PRESENCE_OUT events.
>>>
>>> Diego
>>>
>>>
>>> On Thu, Sep 3, 2009 at 6:40 PM, Jerry Richards <
>>> jerry.richa...@teotech.com> wrote:
>>>
>>>> Does Freeswitch support Presence via SIMPLE protocol?  Can it maintain
>>>> presence?  I presume this would be a SUBSCRIBE/NOTIFY arrangement?
>>>>
>>>> Best Regards,
>>>> Jerry
>>>>
>>>>
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>>>
>>>
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Re: [Freeswitch-users] Digium boards

[Diego Viola]

Sure.

Diego

On Thu, Sep 3, 2009 at 8:49 PM, Pedro Prado  wrote:

>  Hi Diego,
>
> I have a Digium AEX410 to test in laboratory. I think, that it works, not?
>
> Thanks,
> Pedro Prado
>
> --
> Date: Thu, 3 Sep 2009 20:21:01 +
> From: diego.vi...@gmail.com
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Digium boards
>
>
> Any Zaptel-compatible card should work.
>
> Diego
>
> On Thu, Sep 3, 2009 at 8:15 PM, Diego Viola  wrote:
>
> Hi Pedro,
>
> Yes it should, it does.
>
> Diego
>
> 2009/9/3 Pedro Prado 
>
>  Hi,
>
> Does Freeswith work with any Digium boards?
>
> Ps: Sorry for the newbie question.
>
> Thanks,
> Pedro Prado
>
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Re: [Freeswitch-users] Presence Feature

[Diego Viola]

You should try to do something like "event plain all" from the freeswitch
CLI and see all the events that you can use in your apps, etc.

The sky is the limit ;)

Diego

On Thu, Sep 3, 2009 at 8:46 PM, Diego Viola  wrote:

> Hi Luis,
>
> My script simply adds two event hooks for presence (PRESENCE_IN and
> PRESENCE_OUT) and then it passes those events to a method (presence_handler)
> and it prints the "from" which is something like: 1000%192.168.0.2 (the user
> and ip) and the status: which can be Registered or Unregistered.
>
> The script is a daemon, so when you register/unregister a phone you see
> something like this:
>
> *I, [2009-08-29T03:05:20.351467 #3092]  INFO -- : Phone 1000%192.168.0.2 -
> Unregistered
>
> I, [2009-08-29T03:05:20.864802 #3092]  INFO -- : Phone 1000%192.168.0.2 -
> Unregistered
>
> I, [2009-08-29T03:05:20.865763 #3092]  INFO -- : Phone 1000%192.168.0.2 -
> Registered(UDP)
>
> I, [2009-08-29T03:05:21.390911 #3092]  INFO -- : Phone 1000%192.168.0.2 -
> Registered(UDP)
>
> I, [2009-08-29T03:05:21.391308 #3092]  INFO -- : Phone 1000%192.168.0.2 -
> Registered(UDP)
>
> I, [2009-08-29T03:05:21.548433 #3092]  INFO -- : Phone 1000%192.168.0.2 -
> Unregistered*
>
> That's pretty much what it does, but this makes it possible to make apps to
> report the status of phones, etc. I was thinking to send that information on
> a database and then show it on a web interface, with Ajax, etc. on "near
> real time", it would be pretty cool :-).
>
> Let me know if you have any questions or need any help.
>
> Regards,
>
> Diego
>
>
> On Thu, Sep 3, 2009 at 8:29 PM, Luis F Urrea  wrote:
>
>> Diego can you explain a little bit further what your script does.
>>
>> Not really versed into ruby but I certainly interested on these events
>>
>>
>> On Thu, Sep 3, 2009 at 1:30 PM, Diego Viola wrote:
>>
>>> I have made a simple script that catches the PRESENCE IN/OUT events and
>>> passes them to a method to get the information for the phones, you can take
>>> a look at it here.
>>>
>>>
>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/presence/presence.rb?r=14679
>>>
>>> Diego
>>>
>>>
>>> On Thu, Sep 3, 2009 at 7:27 PM, Diego Viola wrote:
>>>
>>>> Not sure if it does via SIMPLE protocol, but if you do "event plain all"
>>>> you should see the PRESENCE_IN and PRESENCE_OUT events.
>>>>
>>>> Diego
>>>>
>>>>
>>>> On Thu, Sep 3, 2009 at 6:40 PM, Jerry Richards <
>>>> jerry.richa...@teotech.com> wrote:
>>>>
>>>>> Does Freeswitch support Presence via SIMPLE protocol?  Can it maintain
>>>>> presence?  I presume this would be a SUBSCRIBE/NOTIFY arrangement?
>>>>>
>>>>> Best Regards,
>>>>> Jerry
>>>>>
>>>>>
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>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>
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Re: [Freeswitch-users] Digium boards

[Diego Viola]

Join #freeswitch at irc.freenode.net if you have any questions, my nick is
"diegoviola" there :).

Diego

On Thu, Sep 3, 2009 at 8:56 PM, Diego Viola  wrote:

> Sure.
>
> Diego
>
>
> On Thu, Sep 3, 2009 at 8:49 PM, Pedro Prado  wrote:
>
>>  Hi Diego,
>>
>> I have a Digium AEX410 to test in laboratory. I think, that it works, not?
>>
>> Thanks,
>> Pedro Prado
>>
>> --
>> Date: Thu, 3 Sep 2009 20:21:01 +
>> From: diego.vi...@gmail.com
>> To: freeswitch-users@lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Digium boards
>>
>>
>> Any Zaptel-compatible card should work.
>>
>> Diego
>>
>> On Thu, Sep 3, 2009 at 8:15 PM, Diego Viola wrote:
>>
>> Hi Pedro,
>>
>> Yes it should, it does.
>>
>> Diego
>>
>> 2009/9/3 Pedro Prado 
>>
>>  Hi,
>>
>> Does Freeswith work with any Digium boards?
>>
>> Ps: Sorry for the newbie question.
>>
>> Thanks,
>> Pedro Prado
>>
>> --
>> Novo Internet Explorer 8: traduza com apenas um clique. Baixe agora, é
>> grátis!<http://brasil.microsoft.com.br/IE8/mergulhe/?utm_source=MSN;Hotmail&utm_medium=Tagline&utm_campaign=IE8>
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>>
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[Freeswitch-users] YAML support as an alternative of XML for configuration

[Diego Viola]

Hi everyone -

I really love this project... I love the design you guys took with FS,
I love the stability, the features, the openness of it, it's the VoIP
platform I always dreamed of it... FreeSWITCH is what Asterisk should
have been.

But I just don't like one thing: XML.

I also don't intend to create a flame war or troll with this thread,
please take this as a suggestion of someone that really wants to use
FS... XML is the only reason why I still stay with Asterisk, I just
can't like XML, and I don't like Asterisk as well, I want FreeSWITCH
stability, features, etc, but I can't like XML, not even if I try to,
it's too slow to use/write, verbose and complex.

As a sysadmin point of view, I prefer parameter=value but I realize
the advantages of XML (data serialization, etc), this is also why I
don't use mod_dialplan_asterisk, because I think it's time for
something better than INI files, but XML isn't for me, I'm a
Linux/UNIX guy and I don't want a GUI either, I love CLI, but I want
something simple to use, read and write, and I think YAML is the best
of the both worlds.

I would like to have more than one option to configure FreeSWITCH,
also, not everyone likes to do the same thing the same way and it
would be really cool if FS gives this freedom to people...

I would really like to see YAML as a configuration option in
FreeSWITCH, give me this any day and I swear I don't toch Asterisk
_ever again_, YAML can be used for data serialization and it's more
simple than XML, Ruby projects like Rails uses it.

Take a look here:

http://yaml.org/spec/1.2/
http://yaml.org/

"YAML™ (rhymes with "camel") is a human-friendly, cross language,
Unicode based data serialization language designed around the common
native data types of agile programming languages. It is broadly useful
for programming needs ranging from configuration files to Internet
messaging to object persistence to data auditing. Together with the
Unicode standard for characters, this specification provides all the
information necessary to understand YAML Version 1.2 and to create
programs that process YAML information. "

Please consider it, I'm also willing to pay for a mod_yaml, other than
this I really love FreeSWITCH and I think it's great, I will switch in
a heartbeat if you guys add support for it ;-)

Best regards,

Diego

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Re: [Freeswitch-users] [Freeswitch-dev] YAML support as an alternative of XML for configuration

[Diego Viola]

Brian,

You said: "If XML is the single reason you won't be using FreeSWITCH
then by all means stick with Asterisk, because we aren't going to beg
you to use FreeSWITCH."

This is sad, you claim that FreeSWITCH is more advanced in terms of
features and innovation than Asterisk, but all I see with your
attitude is that it's more limited and closed-minded.

I'm afraid that I will have to stick with Asterisk, at least they
aren't arrogant and closed-minded when you ask for a feature, all I
want is a more effective way (for me) to configure FreeSWITCH, this is
not going to hurt you or steal your freedom, or change the way you do
things... it's only going to allow me to do things in the way I'm used
to and I like.

This is exactly why I love Free Software & Open Source, because it
gives me the freedom to do things exactly the way I want, but I see I
wont change your mind... so, good luck with FS.

Diego

On Thu, Jun 26, 2008 at 11:15 PM, Brian West <[EMAIL PROTECTED]> wrote:
> Diego,
>I can already tell you Its not going to happen.  The choice was made
> early on and it was the correct one.  It's not something that can be
> changed or modular.  If XML is the single reason you won't be using
> FreeSWITCH then by all means stick with Asterisk, because we aren't
> going to beg you to use FreeSWITCH.
>
> Read this for more info http://www.freeswitch.org/node/123
>
> /b
>
> On Jun 26, 2008, at 9:14 PM, Diego Viola wrote:
>
>> Hi everyone -
>>
>> I really love this project... I love the design you guys took with FS,
>> I love the stability, the features, the openness of it, it's the VoIP
>> platform I always dreamed of it... FreeSWITCH is what Asterisk should
>> have been.
>>
>> But I just don't like one thing: XML.
>>
>> I also don't intend to create a flame war or troll with this thread,
>> please take this as a suggestion of someone that really wants to use
>> FS... XML is the only reason why I still stay with Asterisk, I just
>> can't like XML, and I don't like Asterisk as well, I want FreeSWITCH
>> stability, features, etc, but I can't like XML, not even if I try to,
>> it's too slow to use/write, verbose and complex.
>>
>> As a sysadmin point of view, I prefer parameter=value but I realize
>> the advantages of XML (data serialization, etc), this is also why I
>> don't use mod_dialplan_asterisk, because I think it's time for
>> something better than INI files, but XML isn't for me, I'm a
>> Linux/UNIX guy and I don't want a GUI either, I love CLI, but I want
>> something simple to use, read and write, and I think YAML is the best
>> of the both worlds.
>>
>> I would like to have more than one option to configure FreeSWITCH,
>> also, not everyone likes to do the same thing the same way and it
>> would be really cool if FS gives this freedom to people...
>>
>> I would really like to see YAML as a configuration option in
>> FreeSWITCH, give me this any day and I swear I don't toch Asterisk
>> _ever again_, YAML can be used for data serialization and it's more
>> simple than XML, Ruby projects like Rails uses it.
>>
>> Take a look here:
>>
>> http://yaml.org/spec/1.2/
>> http://yaml.org/
>>
>> "YAML™ (rhymes with "camel") is a human-friendly, cross language,
>> Unicode based data serialization language designed around the common
>> native data types of agile programming languages. It is broadly useful
>> for programming needs ranging from configuration files to Internet
>> messaging to object persistence to data auditing. Together with the
>> Unicode standard for characters, this specification provides all the
>> information necessary to understand YAML Version 1.2 and to create
>> programs that process YAML information. "
>>
>> Please consider it, I'm also willing to pay for a mod_yaml, other than
>> this I really love FreeSWITCH and I think it's great, I will switch in
>> a heartbeat if you guys add support for it ;-)
>>
>> Best regards,
>>
>> Diego
>>
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Re: [Freeswitch-users] [Freeswitch-dev] YAML support as an alternative of XML for configuration

[Diego Viola]

Hi,

I'm sorry for my reaction... I know FreeSWITCH is better compared to
Asterisk, and I will learn it, with XML or not.

I also don't think XML is a bad thing, it's an open standard and I
think I will get used to it.

@Ognjen, when you say:

"You can't be serious that Asterisk/Digium is ok when you ask them for
a feature or anything at all? Tony offered them to rewrite obvious
broken core to make it better, they said no. I tried 1.4 because of
the new features advertised, and I got no sleep beacuse of it for two
weeks. What's the point of polishing something that is without any
doubt broken in core? I don't
understand why are you even looking at FS as an option, if Asterisk is
ok for your needs?"

What do you mean with "broken core" exactly? I know asterisk has
problems but I would like to know what core problems you refer there.
I'm just curious about it.

I'm looking at FS as an alternative to Asterisk because I been having
a lot of problems with Asterisk, and I want a system that works
better, the problems I had are the following:

http://bugs.digium.com/view.php?id=12628
http://bugs.digium.com/view.php?id=12653
http://bugs.digium.com/view.php?id=12810

And I had more than that... and I keep having problems, I want to move
on and have a system that I can trust.

Anyway... I will learn FS and XML, and sorry for my reaction... I know
Brian is a good and cool guy and he is the guy that introduced me to
FS.

Peace out,

Diego

On Sun, Jun 29, 2008 at 10:20 PM, Michael Collins
<[EMAIL PROTECTED]> wrote:
>> This XML argument has gone on since FreeSwitch was unveiled and quite
>> frankly I was one of the people railing on Tony for that decision, but
> as
>> I
>> have continued to look at the core and develop soluitons on this
> platform
>> it
>> has become more and more obvious that I was wrong initially... (wow
>> someone
>> archive tha.. Its the only time you'll see it on a mailing list :P )
>> K
>
>
> So noted!  It would be nice to get an audio recording of you groveling
> for Tony's forgiveness... we could add it to the demo IVR:
>
> "...To hear Ken Rice admit the folly of his ways, please press ..."
>
> :)
>
> -MC
>
> P.S. - Can you record that in UWB mode? =D
>

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Re: [Freeswitch-users] [Freeswitch-dev] YAML support as an alternative of XML for configuration

[Diego Viola]

Read more about it here: http://en.wikipedia.org/wiki/YAML

Diego

On Mon, Jun 30, 2008 at 7:14 PM, Diego Viola <[EMAIL PROTECTED]> wrote:
> This rocks!
>
> Anthony: Thank you so much for this... and sorry for the troubles I
> caused, I do not think you or anyone else in your team is a closed
> minded person... FreeSWITCH is the best fruit of your great talent and
> open mind ;-)
>
> Diego
>
> On Mon, Jun 30, 2008 at 4:43 PM, EdPimentl <[EMAIL PROTECTED]> wrote:
>> Net Net, anyway you want to say... this  is an amazing team.
>>
>> In XML:
>> 
>> Anthony
>> Mike
>> Brian
>> 
>>
>> In YAML:
>>
>> AmazingTeam:
>> project : FreeSwitch
>> projectowner: Anthony
>> projectlead: Mike
>> qa: Brian
>>
>>
>>
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>

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Re: [Freeswitch-users] [Freeswitch-dev] YAML support as an alternative of XML for configuration

[Diego Viola]

This rocks!

Anthony: Thank you so much for this... and sorry for the troubles I
caused, I do not think you or anyone else in your team is a closed
minded person... FreeSWITCH is the best fruit of your great talent and
open mind ;-)

Diego

On Mon, Jun 30, 2008 at 4:43 PM, EdPimentl <[EMAIL PROTECTED]> wrote:
> Net Net, anyway you want to say... this  is an amazing team.
>
> In XML:
> 
> Anthony
> Mike
> Brian
> 
>
> In YAML:
>
> AmazingTeam:
> project : FreeSwitch
> projectowner: Anthony
> projectlead: Mike
> qa: Brian
>
>
>
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[Freeswitch-users] Time out issue and static noise with originate command

[Diego Viola]

Hi all...

I have a user account on FS, sometimes when I register with Zoiper or
X-Lite it works, but after some time it unregisters itself, I'm not
sure why... this didn't happened on Asterisk for me.

Does FS handles NAT automatically? This is how I have it setup:

softphone => NAT (router) => FS

In Asterisk I used to have nat=yes in sip.conf, is there a equivalent
for this in FS?

Why could my softphone be deregistering itself?

This is my user account on FS:

Maitei:/usr/local/freeswitch# cat conf/directory/default/diego.xml

  

  
  


  
  
  
  

  


Another issue... when I use the originate command, with a WAV or GSM
file... all I get is static noise, is there a way that this could
work?, this also works on Asterisk just fine for me:

originate sofia/internal/diego &playback('/root/nin.mp3.wav')

But I get static noise on FS.

Thanks,

Diego

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Re: [Freeswitch-users] Time out issue and static noise with originate command

[Diego Viola]

FreeSWITCH is not behind NAT, only my softphone... looks like it was
only my ISP doing nasty things again.

Thanks for your answer though, it works again... I tried this and
happened with Asterisk too, so I think it was a network issue, but it
works now... thanks.

What about the originate thing, do you know why is not playing the
sound file with playback?

Diego

On Sun, Jul 6, 2008 at 11:41 AM, Brian West <[EMAIL PROTECTED]> wrote:
> So you have FreeSWITCH behind nat? on the "internal" profile?  If
> thats the case you'll need to set the sip-ext-ip and ext-rtp-ip so
> that it can communicate with the outside world properly.
>
> /b
>
> On Jul 5, 2008, at 10:55 PM, Diego Viola wrote:
>
>> softphone => NAT (router) => FS
>
>
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Re: [Freeswitch-users] Time out issue and static noise with originate command

[Diego Viola]

Sure, here I send you the file... all I hear is static noise when I do:

originate sofia/internal/diego &playback('/home/diego/welcometorac.gsm')

My softphone rings but I hear the static noise... from the XML IVR
(ivr.conf.xml) with playback works well...

Thanks,

Diego

On Sun, Jul 6, 2008 at 1:04 PM, Brian West <[EMAIL PROTECTED]> wrote:
> What kind of file is it and can you send me the file?
>
> /b
>
> On Jul 6, 2008, at 11:41 AM, Diego Viola wrote:
>
>>
>> What about the originate thing, do you know why is not playing the
>> sound file with playback?
>
>
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welcometorac.gsm
Description: Binary data
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[Freeswitch-users] IVR XML and phrase macros all in one single file

[Diego Viola]

Hi,

I been playing with the ivr.conf.xml in FS, is there a way to put the
phrase macros inside the ivr.conf.xml, the reason for this is that I
would like to have all my ivr config in one single file.

Something like this: http://pastie.org/228307

Thanks,

Diego

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Re: [Freeswitch-users] Time out issue and static noise with originate command

[Diego Viola]

Uhm, I just originated a call to my landline and it works great... but
when I originate to my cellphone it doesn't work. I get this message:

[EMAIL PROTECTED]> 2008-07-06 20:26:57 [NOTICE] sofia.c:2085
sofia_handle_sip_i_state() Hangup sofia/internal/011595982343565
[CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2008-07-06 20:26:57 [WARNING] mod_sndfile.c:212 sndfile_file_seek()
File is not seekable 2008-07-06 20:26:57 [NOTICE]
switch_core_session.c:753
switch_core_session_thread() Session 5
(sofia/internal/011595982343565) Ended 2008-07-06 20:26:57 [NOTICE]
switch_core_session.c:755
switch_core_session_thread() Close Channel
sofia/internal/011595982343565 [CS_HANGUP]

I'm not sure if is a codec thing, this is what I have in vars.xml:

 
 

This is how I'm originating the call:

[EMAIL PROTECTED]> originate sofia/gateway/teliax/011595982343565
&playback('/home/diego/foo.gsm')

The cellphone rings but I don't hear anything, I tried the same thing
with Asterisk and worked fine... any ideas?

Thanks,

Diego

On Sun, Jul 6, 2008 at 1:17 PM, Diego Viola <[EMAIL PROTECTED]> wrote:
> Sure, here I send you the file... all I hear is static noise when I do:
>
> originate sofia/internal/diego &playback('/home/diego/welcometorac.gsm')
>
> My softphone rings but I hear the static noise... from the XML IVR
> (ivr.conf.xml) with playback works well...
>
> Thanks,
>
> Diego
>
> On Sun, Jul 6, 2008 at 1:04 PM, Brian West <[EMAIL PROTECTED]> wrote:
>> What kind of file is it and can you send me the file?
>>
>> /b
>>
>> On Jul 6, 2008, at 11:41 AM, Diego Viola wrote:
>>
>>>
>>> What about the originate thing, do you know why is not playing the
>>> sound file with playback?
>>
>>
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[Freeswitch-users] help with: [ERR] mod_local_stream.c:266 local_stream_file_open() unknown source moh

[Diego Viola]

Hi

When I dial a conference I get this:

2008-07-07 00:19:45 [ERR] mod_local_stream.c:266
local_stream_file_open() unknown source moh

I think I have all the MOH files:

http://pastebin.com/m6a463dfa
http://pastebin.com/m7f24dce

Maitei:~# grep hold_music /usr/local/freeswitch/conf/vars.xml
  

Sorry if I'm missing something... I'm still very new in FreeSWITCH,
but I'm really enjoying the experience.

Thanks,

Diego

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Re: [Freeswitch-users] help with: [ERR] mod_local_stream.c:266 local_stream_file_open() unknown source moh

[Diego Viola]

I did make hd-moh-install, I also installed all the sound files after
the make install.

I am calling 3232, this one:






  


I don't think I have touched local_stream.conf.xml, this is a recent install.

This is what I have in my local_stream.conf.xml file:


  





  

  





  
  


Shouldn't that be uncommented?

Diego

On Sun, Jul 6, 2008 at 8:31 PM, Brian West <[EMAIL PROTECTED]> wrote:
> You'll have to make sure you have both 8k and 16k files installed.
> Which conference were you calling in the default configs?
>
> make hd-moh-install
>
> Or you have changed local_stream.conf.xml from the defaults.
>
> /b
>
> On Jul 6, 2008, at 7:23 PM, Diego Viola wrote:
>
>> Hi
>>
>> When I dial a conference I get this:
>>
>> 2008-07-07 00:19:45 [ERR] mod_local_stream.c:266
>> local_stream_file_open() unknown source moh
>>
>> I think I have all the MOH files:
>>
>> http://pastebin.com/m6a463dfa
>> http://pastebin.com/m7f24dce
>>
>> Maitei:~# grep hold_music /usr/local/freeswitch/conf/vars.xml
>>  
>>
>> Sorry if I'm missing something... I'm still very new in FreeSWITCH,
>> but I'm really enjoying the experience.
>>
>> Thanks,
>>
>> Diego
>>
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Re: [Freeswitch-users] help with: [ERR] mod_local_stream.c:266 local_stream_file_open() unknown source moh

[Diego Viola]

Yes, I uncommented that... it works now, thanks Brian ;-)

Diego

On Sun, Jul 6, 2008 at 8:49 PM, Diego Viola <[EMAIL PROTECTED]> wrote:
> I did make hd-moh-install, I also installed all the sound files after
> the make install.
>
> I am calling 3232, this one:
>
>
>
>
>
>
>  
>
>
> I don't think I have touched local_stream.conf.xml, this is a recent install.
>
> This is what I have in my local_stream.conf.xml file:
>
> 
>  
>
>
>
>
>
>  
>
>  
>
>
>
>
>
>  
>  
> 
>
> Shouldn't that be uncommented?
>
> Diego
>
> On Sun, Jul 6, 2008 at 8:31 PM, Brian West <[EMAIL PROTECTED]> wrote:
>> You'll have to make sure you have both 8k and 16k files installed.
>> Which conference were you calling in the default configs?
>>
>> make hd-moh-install
>>
>> Or you have changed local_stream.conf.xml from the defaults.
>>
>> /b
>>
>> On Jul 6, 2008, at 7:23 PM, Diego Viola wrote:
>>
>>> Hi
>>>
>>> When I dial a conference I get this:
>>>
>>> 2008-07-07 00:19:45 [ERR] mod_local_stream.c:266
>>> local_stream_file_open() unknown source moh
>>>
>>> I think I have all the MOH files:
>>>
>>> http://pastebin.com/m6a463dfa
>>> http://pastebin.com/m7f24dce
>>>
>>> Maitei:~# grep hold_music /usr/local/freeswitch/conf/vars.xml
>>>  
>>>
>>> Sorry if I'm missing something... I'm still very new in FreeSWITCH,
>>> but I'm really enjoying the experience.
>>>
>>> Thanks,
>>>
>>> Diego
>>>
>>> ___
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>>
>>
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Re: [Freeswitch-users] help with: [ERR] mod_local_stream.c:266 local_stream_file_open() unknown source moh

[Diego Viola]

This is very cool, thanks Brian.

I get it now, FreeSWITCH rocks! =D

Diego

On Sun, Jul 6, 2008 at 8:55 PM, Brian West <[EMAIL PROTECTED]> wrote:
> Then you need to install the 32k files and modify the
> local_stream.conf.xml to enable the 32k stream.
>
> make uhd-moh-install
>
> You have very little reason to use 32k audio not much if anything can
> use it yet.  Except FS :P
>
> /b
>
>
>
> On Jul 6, 2008, at 7:49 PM, Diego Viola wrote:
>
>> I did make hd-moh-install, I also installed all the sound files after
>> the make install.
>>
>> I am calling 3232, this one:
>>
>>
>>
>>
>>
>>
>>  
>>
>
>
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[Freeswitch-users] FreeSWITCH and Amazon EC2 port problem

[Diego Viola]

Hi

I have FreeSWITCH 1.0.0 installed on Amazon EC2, today was working
very well... but after I re-installed FS I'm getting some problem with
ports.

I did followed this guide and opened the required ports:
http://wiki.freeswitch.org/wiki/Amazon_ec2

The problem is when I run FS, I see that the normal ports are opened:


[EMAIL PROTECTED]> sofia status
API CALL [sofia(status)] output:
 Name  Type
Data  State
=
 external   profile
sip:[EMAIL PROTECTED]:5080  RUNNING (0)
  nat   profile
sip:[EMAIL PROTECTED]:5070  RUNNING (0)
  default alias
internal  ALIASED
 internal   profile
sip:[EMAIL PROTECTED]:5060 RUNNING (0)
ec2-67-202-9-134.compute-1.amazonaws.com  alias
   internal  ALIASED
 outbound alias
external  ALIASED
=
3 profiles 3 aliases


But when I do a netstat -a, I see that 5060 is not opened... there is
a 5070 port opened though, but my softphone (X-Lite) doesn't
register... what could be wrong?

Thanks,

Diego

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Re: [Freeswitch-users] FreeSWITCH and Amazon EC2 port problem

[Diego Viola]

Output of "sofia status":

[EMAIL PROTECTED]> sofia status
API CALL [sofia(status)] output:
 Name  Type
Data  State
=
 external   profile
sip:[EMAIL PROTECTED]:5080  RUNNING (0)
  nat   profile
sip:[EMAIL PROTECTED]:5070  RUNNING (0)
  default alias
internal  ALIASED
 internal   profile
sip:[EMAIL PROTECTED]:5060 RUNNING (0)
ec2-67-202-9-134.compute-1.amazonaws.com  alias
   internal  ALIASED
 outbound alias
external  ALIASED
=
3 profiles 3 aliases

Output of netstat -a:

Etch:/usr/local/freeswitch# netstat -a
Active Internet connections (servers and established)
Proto Recv-Q Send-Q Local Address   Foreign Address State
tcp0  0 ip-10-251-203-143.e:sip *:* LISTEN
tcp0  0 ip-10-251-203-143.:5070 *:* LISTEN
tcp0  0 localhost:zope-ftp  *:* LISTEN
tcp0  0 *:ssh   *:* LISTEN
tcp0  0 ip-10-251-203-143.:5080 *:* LISTEN
tcp0 48 ip-10-251-203-143.e:ssh adsl-132-64.click:63370 ESTABLISHED
udp0  0 ip-10-251-203-143.e:sip *:*
udp0  0 *:bootpc*:*
udp0  0 ip-10-251-203-143.:5070 *:*
udp0  0 ip-10-251-203-143.:5080 *:*
Active UNIX domain sockets (servers and established)
Proto RefCnt Flags   Type   State I-Node Path
unix  4  [ ] DGRAM1093 /dev/log
unix  3  [ ] STREAM CONNECTED 167068
unix  3  [ ] STREAM CONNECTED 167067
unix  3  [ ] STREAM CONNECTED 167063
unix  3  [ ] STREAM CONNECTED 167062
unix  3  [ ] STREAM CONNECTED 167001
unix  3  [ ] STREAM CONNECTED 167000
unix  3  [ ] STREAM CONNECTED 166988
unix  3  [ ] STREAM CONNECTED 166987
unix  3  [ ] STREAM CONNECTED 166983
unix  3  [ ] STREAM CONNECTED 166982
unix  3  [ ] STREAM CONNECTED 166968
unix  3  [ ] STREAM CONNECTED 166967
unix  2  [ ] DGRAM156291
unix  2  [ ] DGRAM1108
Etch:/usr/local/freeswitch#

FreeSWITCH says that port 5060 is open but netstat -a doesn't show
it... and X-Lite wont register, what could be wrong?

Diego

On Sat, Jul 12, 2008 at 2:29 AM, Diego Viola <[EMAIL PROTECTED]> wrote:
> Hi
>
> I have FreeSWITCH 1.0.0 installed on Amazon EC2, today was working
> very well... but after I re-installed FS I'm getting some problem with
> ports.
>
> I did followed this guide and opened the required ports:
> http://wiki.freeswitch.org/wiki/Amazon_ec2
>
> The problem is when I run FS, I see that the normal ports are opened:
>
>
> [EMAIL PROTECTED]> sofia status
> API CALL [sofia(status)] output:
> Name  Type
> Data  State
> =
> external   profile
> sip:[EMAIL PROTECTED]:5080  RUNNING (0)
>  nat   profile
> sip:[EMAIL PROTECTED]:5070  RUNNING (0)
>  default alias
> internal  ALIASED
> internal   profile
> sip:[EMAIL PROTECTED]:5060 RUNNING (0)
> ec2-67-202-9-134.compute-1.amazonaws.com  alias
>   internal  ALIASED
> outbound alias
> external  ALIASED
> =
> 3 profiles 3 aliases
>
>
> But when I do a netstat -a, I see that 5060 is not opened... there is
> a 5070 port opened though, but my softphone (X-Lite) doesn't
> register... what could be wrong?
>
> Thanks,
>
> Diego
>

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Re: [Freeswitch-users] FreeSWITCH and Amazon EC2 port problem

[Diego Viola]

I got it, thanks Brian.

Etch:/usr/local/freeswitch# netstat -na
Active Internet connections (servers and established)
Proto Recv-Q Send-Q Local Address   Foreign Address State
tcp0  0 10.251.203.143:5060 0.0.0.0:*   LISTEN
tcp0  0 10.251.203.143:5070 0.0.0.0:*   LISTEN
tcp0  0 127.0.0.1:8021  0.0.0.0:*   LISTEN
tcp0  0 0.0.0.0:22  0.0.0.0:*   LISTEN
tcp0  0 10.251.203.143:5080 0.0.0.0:*   LISTEN
tcp0 96 10.251.203.143:22   190.52.132.64:61347 ESTABLISHED
udp0  0 10.251.203.143:5060 0.0.0.0:*
udp0  0 0.0.0.0:68  0.0.0.0:*
udp0  0 10.251.203.143:5070 0.0.0.0:*
udp0  0 10.251.203.143:5080 0.0.0.0:*
Active UNIX domain sockets (servers and established)
Proto RefCnt Flags   Type   State I-Node Path
unix  4  [ ] DGRAM1093 /dev/log
unix  3  [ ] STREAM CONNECTED 168159
unix  3  [ ] STREAM CONNECTED 168158
unix  3  [ ] STREAM CONNECTED 168153
unix  3  [ ] STREAM CONNECTED 168152
unix  3  [ ] STREAM CONNECTED 168147
unix  3  [ ] STREAM CONNECTED 168146
unix  3  [ ] STREAM CONNECTED 168142
unix  3  [ ] STREAM CONNECTED 168141
unix  3  [ ] STREAM CONNECTED 168132
unix  3  [ ] STREAM CONNECTED 168131
unix  3  [ ] STREAM CONNECTED 168127
unix  3  [ ] STREAM CONNECTED 168126
unix  2  [ ] DGRAM156291
unix  2  [ ] DGRAM1108
Etch:/usr/local/freeswitch#


On Sat, Jul 12, 2008 at 9:13 AM, Brian West <[EMAIL PROTECTED]> wrote:
> you'll need to do netstat -na but this one is :sip which means :5060
>
> /b
>
> On Jul 12, 2008, at 1:30 AM, Diego Viola wrote:
>
>> tcp0  0 ip-10-251-203-143.e:sip *:*
>> LISTEN
>
>
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[Freeswitch-users] FreeSWITCH and NAT problem

[Diego Viola]

Hi,

I'm running FreeSWITCH on a server that has no NAT in between... the
server is on Slicehost (slicehost.com), a VPS.

I'm trying to register my softphone (Zoiper and X-Lite) from my
computer, my computer is behind a router (NAT). So the scenario is the
following:

Softphone (Zoiper or X-Lite) -> router (NAT) -> INTERNET -> FS server
(no nat -- public ip address).

The problem is that I can't make it register, sometimes it will
register and sometimes it wont, this problem is randomly.

This is what I have in vars.xml:

  

I'm not sure if this has something to be with STUN or NAT, it works
fine with Asterisk... in Asterisk I have nat=yes in my SIP user
(sip.conf), is there a equivalent of nat=yes in FS that I can use for
my user?

Thanks,

Diego

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Re: [Freeswitch-users] Some little help with regexp and extensions

[Diego Viola]

mercutioviz in #freeswitch already solved it for me, a big thanks for you =D

The trick was to use ^9(\d{10})$. I need to get off and learn a bit of
regexp ;-)

Here is the log in case if someone wants to read it, maybe it will help some:

19:39 < mercutioviz> diegoviola: i've had a fair amount of success
with regexps in the dialplan
19:40 < mercutioviz> what are you hoping to accomplish?
19:40 < mercutioviz> gimme the plain english and i'll help u with a regexp
19:41 < diegoviola> mercutioviz: well i had a 9(\d+)$ for dialing out
through my gateway and at the same time i have the default extensions
for echo (9996) etc... and my dial out regexp extension also matches
9996
19:41 < mercutioviz> yep, classic problem
19:42 < mercutioviz> how many digits will the caller dial after
dialing 9? like 9 12345678
19:42 < diegoviola> normally is 9-1-number
19:42 < mercutioviz> no prob
19:42 < diegoviola> that's what i was using
19:43 < diegoviola> wait
19:43 < diegoviola> ^(\d+)$ works fine for me
19:43 < diegoviola> no need for the 9 first i think
19:43 < mercutioviz> try this: ^9(\d{10,})$
19:43 < mercutioviz> it depends on how fancy you want to get
19:44 < mercutioviz> example: 9+phone number goes out one provider,
8+phone number goes out another provider
19:44 < mercutioviz> etc.
19:44 < mercutioviz> the above regexp says "Match if I see 9 plus at
least ten consecutive digits"
19:44 < diegoviola> with ^(\d+)$ i can't dial 3030, etc
19:44 < mercutioviz> that way 9996 won't match
19:45 < mercutioviz> the problem with ^(\d+)$ is that it will match
*any* consecutive string of digits
19:45 < diegoviola> interesting

Thanks, FS rocks =D

Diego

 Jul 30, 2008 at 5:29 PM, Diego Viola <[EMAIL PROTECTED]> wrote:
> Hi everyone,
>
> I have this in my default dialplan for dialing out:
>
> 
>  
>
>  
> 
>
> It works fairly well... but when I try to dial other extensions that
> start with "9", like 9996 (echo extension) I get this:
> http://pastebin.freeswitch.org/5026 and in my softphone I get "service
> or option unavailable".
>
> If I remove the "outbound" extension works.
>
> I know it might be something simple... but I'm not very familiar with
> regular expressions yet, so if you can help me with this I will be
> really thankful.
>
> Thanks everyone... FreeSWITCH rocks =D
>
> Diego
>

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[Freeswitch-users] Some little help with regexp and extensions

[Diego Viola]

Hi everyone,

I have this in my default dialplan for dialing out:


  

  


It works fairly well... but when I try to dial other extensions that
start with "9", like 9996 (echo extension) I get this:
http://pastebin.freeswitch.org/5026 and in my softphone I get "service
or option unavailable".

If I remove the "outbound" extension works.

I know it might be something simple... but I'm not very familiar with
regular expressions yet, so if you can help me with this I will be
really thankful.

Thanks everyone... FreeSWITCH rocks =D

Diego

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Re: [Freeswitch-users] How to reconnect to FS ?

[Diego Viola]

Just as a suggestion, it would be nice a -r option in the freeswitch
binary, could this be done in the future? =D

Diego

On Thu, Jul 31, 2008 at 2:58 PM, Brian West <[EMAIL PROTECTED]> wrote:
> Also make sure the FreeSWITCH/ dir is in your site_perl dir
>
> /b
>
> On Jul 31, 2008, at 1:54 PM, unknown wrote:
>
>> freeswitch# perl fs.pl
>> FreeSWITCH>
>>
>> That's it.
>>
>> Chris
>
> Brian West
> sip:[EMAIL PROTECTED]
>
>
>
>
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Re: [Freeswitch-users] How to reconnect to FS ?

[Diego Viola]

I know there is fs.pl, and it's indeed useful, but an -r option in the
binary itself would be nicer ;-)

Just as a suggestion, don't take it bad =D.

FS rocks!

Diego

On Thu, Jul 31, 2008 at 3:57 PM, Henk Oegema <[EMAIL PROTECTED]> wrote:
> On Thursday 31 July 2008 20:54:53 unknown wrote:
>> freeswitch# perl fs.pl
>> FreeSWITCH>
>>
>> That's it.
>
>
> :-)
>
> Henk
>
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Re: [Freeswitch-users] How to reconnect to FS ?

[Diego Viola]

It would be cool also if you could re-attach FS from internet and from
the local host, with -r (resume) -h (host) -p (port). So you could
re-attach not just locally but from any host.

Just a though ;-)

Diego

On Thu, Jul 31, 2008 at 6:41 PM, Ashutosh <[EMAIL PROTECTED]> wrote:
> I wondered if attaching FS to a a "screen" and in bg mode would be good
> workaround, so that one can reattach to that screen from any other shell
> later ?
>
> Regards,
> ashutosh
>
> On Fri, Aug 1, 2008 at 1:32 AM, Diego Viola <[EMAIL PROTECTED]> wrote:
>>
>> I know there is fs.pl, and it's indeed useful, but an -r option in the
>> binary itself would be nicer ;-)
>>
>> Just as a suggestion, don't take it bad =D.
>>
>> FS rocks!
>>
>> Diego
>>
>> On Thu, Jul 31, 2008 at 3:57 PM, Henk Oegema <[EMAIL PROTECTED]>
>> wrote:
>> > On Thursday 31 July 2008 20:54:53 unknown wrote:
>> >> freeswitch# perl fs.pl
>> >> FreeSWITCH>
>> >>
>> >> That's it.
>> >
>> >
>> > :-)
>> >
>> > Henk
>> >
>> > ___
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[Freeswitch-users] Help with call forwarding

[Diego Viola]

Hi everyone,

I'm trying to do call forwarding in FS, this is what I have in my
default dialplan:











Whenever I dial 1, I get a NO_ROUTE_DESTINATION error message
because the UA is not registered.  If I add NO_ROUTE_DESTINATION to
the continue_on_fail line, it doesn't work too.

I need it to pass to the next call even if the UA (scot.clausing) is
not registered, or if there is no route to destination.

Thanks,

Diego

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Re: [Freeswitch-users] Help with call forwarding

[Diego Viola]

I just added NO_ROUTE_DESTINATION in continue_on_fail and it works now.

If I remove the 
works as well =D

Thanks everyone, FS rocks! =D

Diego

On Wed, Aug 13, 2008 at 4:04 PM, Diego Viola <[EMAIL PROTECTED]> wrote:
> Hi everyone,
>
> I'm trying to do call forwarding in FS, this is what I have in my
> default dialplan:
>
> 
> 
> 
> 
>  data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER"/>
> 
> 
> 
> 
>
> Whenever I dial 1, I get a NO_ROUTE_DESTINATION error message
> because the UA is not registered.  If I add NO_ROUTE_DESTINATION to
> the continue_on_fail line, it doesn't work too.
>
> I need it to pass to the next call even if the UA (scot.clausing) is
> not registered, or if there is no route to destination.
>
> Thanks,
>
> Diego
>

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Re: [Freeswitch-users] Request for change

[Diego Viola]

Sunil,

You are very wrong.

Anthony Minessale does listen to his users, in fact... A LOT MORE than
what any Asterisk developer EVER DID or will.

FreeSWITCH people is made of people who does love what they do... they
wanted something better than Asterisk and they did it, that's pure
love. We don't hate the asterisk community, but we love FreeSWITCH.

Search for the YAML thing that I suggested once, Anthony added that
feature in only a few hours... he indeed does listen to his users and
he does a great job with FreeSWITCH.

Get your facts straight before trolling like that again.

Diego

On Sat, Aug 30, 2008 at 6:38 AM, Sunil Singh <[EMAIL PROTECTED]> wrote:
> Hi,
>
> After going through the documents, I feel that the documents assume that the
> guy who is going to use freeswitch is well versed with asterisk or is an
> asterisk user who is going to switch to freeswitch from asterisk. Its not
> from the point that somebody from scratch can appreciate it for its design
> or any other thing.
>
> I appreciate the point of consideration for a large community, but for
> freeswitch to be the best it has to wipe out that they know asterisk. They
> should believe more on its users rather than what asterisk is doing. Once a
> basic level is reached than it should be user driven rather than asterisk
> driven. If we compete with them in mind than eventually we will mess up.
>
> To be honest the beauty of asterisk lies in its simplicity. I was reading
> freeswitch for hours and still struggling. Whereas in asterisk I started and
> within hours I am up and running.
>
> I also realised that in asterisk I didn't find a document talking about
> freeswitch but in freeswitch its only about, with asterisk in mind. Whether
> they are documents or working.
>
> Comparisions will not allow to grow beyond a limit, we will always be living
> in a psycho complex, leaving us with pain, agaony and anger.
>
> We should love what we are doing, and we want to make a universal platform
> thats it. No point in comparision and running for a race. Most of the open
> source are started by technology lovers that's it. They don't create a
> technology to compete, they want to create a new technology and that's why I
> started using freeswitch that its going to create a new revolution which
> asterisk has lost because of monetary considerations.
>
> So I request to please love what we are doing and when ever we are making a
> document think that its for a novice.
>
>
> I request those who are driving it, please listen to your users more. If you
> have started from scratch than let's some beauty come out of it, a
> disruptive way of working. Like Apache, mysql ..  Ease of use is the
> main thing for a starter or anybody else. Even it may be a best but if its
> not simple than people are not going to use it and too much mention of
> asterisk in the doc will end up people trying asterisk more rather than
> using freeswitch.
>
> The guys whom we are targetting today may not be there, there will be a new
> generation and we should drive it with them in mind.
>
> I hope we all give it a thought and think from a novice point of view.
>
> Be a BOLT win with a self confidence and create only records.
>
> regards,
> Sunil
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
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Re: [Freeswitch-users] Open g729 g723 codec, any expierence

[Diego Viola]

I would like to have this, since I live in a country where patents
don't affect us.

Diego

On Mon, Sep 1, 2008 at 11:00 PM, Lee JJ <[EMAIL PROTECTED]> wrote:
> Hello :
>
> Anyone have tried to compile open codec into FW ?
> http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
>
> thanks
>
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Re: [Freeswitch-users] Open g729 g723 codec, any expierence

[Diego Viola]

I live in Paraguay, doesn't that only affects in the U.S? Correct me
if I'm wrong, but I guess the patents issue is only in the U.S.

Diego

On Tue, Sep 2, 2008 at 8:02 AM, Steve Underwood <[EMAIL PROTECTED]> wrote:
> Brian West wrote:
>> I would watch a blanket statement like that.  You would be surprised
>> to find out how far those patents reach.
>>
>> /b
>>
>> On Sep 1, 2008, at 11:19 PM, Diego Viola wrote:
>>
>>
>>> I would like to have this, since I live in a country where patents
>>> don't affect us.
>>>
>>> Diego
>>>
> Don't be so judgemental. He might be part of an Antarctic expedition, or
> spend his entire life afloat outside territorial waters. :-)
>
> Steve
>
>
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Re: [Freeswitch-users] SIP, NAT and Amazon EC2

[Diego Viola]

in vars.xml

On Sat, Sep 6, 2008 at 1:21 AM, Diego Viola <[EMAIL PROTECTED]> wrote:
> Try changing these lines and put your ip instead, that worked for me.
>
>  
>  
>
> Diego
>
> On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown <[EMAIL PROTECTED]> wrote:
>> Ive Tried the following with no success:
>>
>> internal.xml
>>
>>
>>
>>
>> Ive also tried just changing the vars.xml file.  I also tried forwarding the 
>> incoming rtp connections to my test pc.  all with no audio success.  I am 
>> sure im doing something wrong I jsut cant find it.
>>
>> I found another suggestion on the wiki of creating a "double nat" that 
>> listens on 5090.  That didnt change anything either, here is that 
>> information:
>>
>> 
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>> From: "Brian West" <[EMAIL PROTECTED]>
>> Sent: Friday, September 5, 2008 6:57pm
>> To: freeswitch-users@lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
>>
>> You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml
>> profile on ec2 duplicate them from the external profile.
>>
>> /b
>>
>> On Sep 5, 2008, at 8:47 PM, Damon Brown wrote:
>>
>>> Yes, I have all of the valid posts open on my security group
>>> -d
>>
>> Brian West
>> sip:[EMAIL PROTECTED]
>>
>>
>>
>>
>>
>>
>>
>> ___
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>>
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Re: [Freeswitch-users] SIP, NAT and Amazon EC2

[Diego Viola]

Try changing these lines and put your ip instead, that worked for me.

  
  

Diego

On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown <[EMAIL PROTECTED]> wrote:
> Ive Tried the following with no success:
>
> internal.xml
>
>
>
>
> Ive also tried just changing the vars.xml file.  I also tried forwarding the 
> incoming rtp connections to my test pc.  all with no audio success.  I am 
> sure im doing something wrong I jsut cant find it.
>
> I found another suggestion on the wiki of creating a "double nat" that 
> listens on 5090.  That didnt change anything either, here is that information:
>
> 
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> From: "Brian West" <[EMAIL PROTECTED]>
> Sent: Friday, September 5, 2008 6:57pm
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
>
> You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml
> profile on ec2 duplicate them from the external profile.
>
> /b
>
> On Sep 5, 2008, at 8:47 PM, Damon Brown wrote:
>
>> Yes, I have all of the valid posts open on my security group
>> -d
>
> Brian West
> sip:[EMAIL PROTECTED]
>
>
>
>
>
>
>
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Re: [Freeswitch-users] registration problem

[Diego Viola]

Is NAT involved in your network?

Diego

On Wed, Sep 10, 2008 at 3:42 AM, Gayatri Kulkarni
<[EMAIL PROTECTED]> wrote:
> Hi Guys,
>
> I have FS installed on 58.86.117.43
> I have the X-lite on 68.86.117.44
> The X-Lite cannot register to the FS - can you guess why?
> SIP account settings for Xlite:
>
> Username: 1002
> Password: 1234
> Auth user name: 1002
> domain: 58.86.117.43
>
> Topology:
> IP address: Discover at runtime
> STUN server: Use specific - 58.86.117.43
>
> I am using a default extension of FS, I dont think i need a gateway here -
> am i correct?
> Please reply soon.
>
> Appreciate all your help,
>
> --
> Regards,
> Gayatri Kulkarni
>
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Re: [Freeswitch-users] registration problem

[Diego Viola]

I don't have much experience with FS behind NAT, but check this out,
maybe it will help:

http://wiki.freeswitch.org/wiki/NAT
http://wiki.freeswitch.org/wiki/NAT_Traversal
http://wiki.freeswitch.org/wiki/External_profile
http://wiki.freeswitch.org/wiki/NAT_Traversal

Diego

On Wed, Sep 10, 2008 at 7:10 AM, Gayatri Kulkarni
<[EMAIL PROTECTED]> wrote:
> i think so
>
> On Wed, Sep 10, 2008 at 3:09 PM, Diego Viola <[EMAIL PROTECTED]> wrote:
>>
>> Is NAT involved in your network?
>>
>> Diego
>>
>> On Wed, Sep 10, 2008 at 3:42 AM, Gayatri Kulkarni
>> <[EMAIL PROTECTED]> wrote:
>> > Hi Guys,
>> >
>> > I have FS installed on 58.86.117.43
>> > I have the X-lite on 68.86.117.44
>> > The X-Lite cannot register to the FS - can you guess why?
>> > SIP account settings for Xlite:
>> >
>> > Username: 1002
>> > Password: 1234
>> > Auth user name: 1002
>> > domain: 58.86.117.43
>> >
>> > Topology:
>> > IP address: Discover at runtime
>> > STUN server: Use specific - 58.86.117.43
>> >
>> > I am using a default extension of FS, I dont think i need a gateway here
>> > -
>> > am i correct?
>> > Please reply soon.
>> >
>> > Appreciate all your help,
>> >
>> > --
>> > Regards,
>> > Gayatri Kulkarni
>> >
>> > ___
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>> >
>> >
>>
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>
>
>
> --
> Regards,
> Gayatri Kulkarni
>
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