Re: [Freeswitch-users] Freeswitch Documentation

2009-12-14 Thread Nagalenoj H.
Great job. You would hear lots of thanks, if you could hear the entire
world.! Thanks.

Regards,
Nagalenoj H.

On Mon, Dec 14, 2009 at 6:57 AM, David V. Fansler
wrote:

>  Final Count was just over 900 files.  At the moment I am putting them in
> a logical order – as best I can tell with my limited experience – combining
> chapters and providing links from the table of contents to sections in each
> chapter.
>
>
>
> Then I will go back in retarget all the hyperlinks to point to the document
> rather than the wiki site.  This may take a few hours J
>
>
>
> David
>
>
>
> David V. Fansler
>
> s/v Annabelle
>
> dfans...@dv-fansler.com
>
> www.dv-fansler.com
>
>
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Robert
> Clayton
> *Sent:* Saturday, December 12, 2009 10:42 PM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Freeswitch Documentation
>
>
>
> David,
>
> Thanks for your hard work.
> Maybe more organization will make the areas needing substance or
> explanation  more obvious.
>
> Bob
>
> On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler <
> dfans...@dv-fansler.com> wrote:
>
> I am new to FreeSWITCH (ok a month old) and am still learning as hard as I
> can.  In the recent talk about documentation, I had noticed that finding
> documentation on the FreeSWITCH wiki was a bit of a chore.  So I decided to
> download all the wiki pages and put them in some sort of logical order.  No
> I am not finished yet, but does anyone else out there realize how many wiki
> pages there are on FreeSWITCH's site?  So far I am up to 767 pages and have
> another 100 to go.
>
> I had no idea what I was getting into!
>
> David
>
> David V. Fansler
> s/v Annabelle
> dfans...@dv-fansler.com
> www.dv-fansler.com
>
>
>
>
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Re: [Freeswitch-users] UUID of the newly originated call?

2009-10-23 Thread Nagalenoj

It is working fine in the current version(15203) of SVN. Thanks.


Anthony Minessale-2 wrote:
> 
> please update to svn trunk with "make current" and try again.
> 
> 
> On Thu, Oct 22, 2009 at 4:08 AM, Nagalenoj  wrote:
> 
>>
>> I'm using 'FreeSWITCH Version 1.0.trunk (15106)'.
>>
>>
>> Anthony Minessale-2 wrote:
>> >
>> > which revision of FS are you using?
>> >
>> > On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. 
>> wrote:
>> >
>> >> I've tried with origination_uuid.
>> >>
>> >> First, I tried with SIP and my program executes successfully as what I
>> >> expected. This program initiates a new call when a call comes and let
>> the
>> >> new call to eavesdrop the landed call.
>> >> When, I experimented with PRI(openzap), I'm facing the following
>> error.
>> >> And, I am unable to make any calls(even from CLI). It is reporting the
>> >> same
>> >> error for the subsequent calls.
>> >> Error:2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760
>> Session
>> >> for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]
>> >>
>> >> Full log is in http://pastebin.freeswitch.org/10780
>> >> My script is here, http://pastebin.freeswitch.org/10781
>> >>
>> >> What is this error for and how to avoid this?
>> >>
>> >> Is there any other way to get the uuid of the originated call except
>> >> explicitly defining(origination_uuid).?!
>> >>
>> >> --
>> >> Regards,
>> >> Nagalenoj H.
>> >>
>> >> ___
>> >> FreeSWITCH-users mailing list
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>> >>
>> >>
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_miness...@hotmail.com
>> <
>> msn%3aanthony_miness...@hotmail.com
>> >
>> >
>> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
>> 
>> >
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:8...@conference.freeswitch.org
>> <
>> sip%3a...@conference.freeswitch.org
>> >
>> > iax:gu...@conference.freeswitch.org/888
>> >
>> googletalk:conf+...@conference.freeswitch.org
>> 
>> >
>> > pstn:213-799-1400
>> >
>> > ___
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>> >
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>>
>> --
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>>
>>
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> 
> 
> 
> -- 
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Re: [Freeswitch-users] UUID of the newly originated call?

2009-10-22 Thread Nagalenoj

I'm using 'FreeSWITCH Version 1.0.trunk (15106)'.


Anthony Minessale-2 wrote:
> 
> which revision of FS are you using?
> 
> On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H.  wrote:
> 
>> I've tried with origination_uuid.
>>
>> First, I tried with SIP and my program executes successfully as what I
>> expected. This program initiates a new call when a call comes and let the
>> new call to eavesdrop the landed call.
>> When, I experimented with PRI(openzap), I'm facing the following error.
>> And, I am unable to make any calls(even from CLI). It is reporting the
>> same
>> error for the subsequent calls.
>> Error:2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session
>> for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]
>>
>> Full log is in http://pastebin.freeswitch.org/10780
>> My script is here, http://pastebin.freeswitch.org/10781
>>
>> What is this error for and how to avoid this?
>>
>> Is there any other way to get the uuid of the originated call except
>> explicitly defining(origination_uuid).?!
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> ___
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_miness...@hotmail.com 
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
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> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
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> 
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Re: [Freeswitch-users] UUID of the newly originated call?

2009-10-20 Thread Nagalenoj H.
I've tried with origination_uuid.

First, I tried with SIP and my program executes successfully as what I
expected. This program initiates a new call when a call comes and let the
new call to eavesdrop the landed call.
When, I experimented with PRI(openzap), I'm facing the following error. And,
I am unable to make any calls(even from CLI). It is reporting the same error
for the subsequent calls.
Error:2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session
for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9]

Full log is in http://pastebin.freeswitch.org/10780
My script is here, http://pastebin.freeswitch.org/10781

What is this error for and how to avoid this?

Is there any other way to get the uuid of the originated call except
explicitly defining(origination_uuid).?!

-- 
Regards,
Nagalenoj H.
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[Freeswitch-users] Bridge in event socket - async mode

2009-10-14 Thread Nagalenoj H.
Hi,

I'm using event outboud socket(perl) in async mode.

Scenario where I face problem:
When a call comes to an extension(1000), my program will play some message
to the user, and get some DTMF.
I'll get the DTMF event(as I'm in async mode) and store the digits in a
variable.

When he presses "#", I call $con->execute("bridge","user/$dtmf"), to bridge
him to the dialed number.

It is bridging sometimes, and sometimes it is not bridging.

Is it the problem with async mode??
I also set $con->setEventLock("true"), before executing the bridge. But
still the problem persists.

I've seen the logs too, I didn't find the bridge application getting
executed.
What makes the problem here..??

The freeswitch log, when bridge is happening:
http://pastebin.freeswitch.org/10704

The freeswitch log, when bridging is not happening:
http://pastebin.freeswitch.org/10705

Code:

 # some statements

if($name eq "CHANNEL_BRIDGE") { # checking for events
.
}
elsif ($name eq "DTMF") {
   my $digit = $e->getHeader("dtmf-digit");

   if($digit eq '#') {
print "Going to bridge\n";
    $con->execute("bridge","user/$dtmf_digit"); 
   }
   else {
$dtmf_digit.=$digit;
   }
}


-- 
Regards,
Nagalenoj H.
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Re: [Freeswitch-users] Re corded file as voicemail.

2009-10-11 Thread Nagalenoj

Is it possible to play a stereo file ?!!! whats the conclusion.?!


Brian West-3 wrote:
> 
> Please open a jira please this did work but a recent change in  
> switch_core_codec caused this to appear I usually test this regularly  
> but haven't run thru a full run of tests lately.
> /b
> 
> On Oct 11, 2009, at 9:27 PM, Seven Du wrote:
> 
>> I set to true because brian said it can play stereo files but no  
>> lucky for me.
> 
> 
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Re: [Freeswitch-users] Re corded file as voicemail.

2009-10-09 Thread Nagalenoj

No, When I do voicemail_inject it is reporting the following error.

'Stereo is currently not supported, please downsample to mono.'

Any other way to do this.??


Michael Jerris wrote:
> 
> http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject
> 
> On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote:
> 
>>
>> Is it possible to treat a recorded voice as voice mail?
>>
>> Assume that, I've recorded a conversation and I want this recorded  
>> file to
>> be treated like voicemail. So, I could check it like voicemail!!
> 
> 
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[Freeswitch-users] UUID of the newly originated call?

2009-10-05 Thread Nagalenoj

Dear friends,
 I am trying with ESL outbound socket. I'm trying to make a call when I
receive ANSWER event. Now, I would want to do something like,
 * Receive the events only for this uuid - I have done by registering
all events and filtering only for this uuid($uuid).
 * If it is CHANNEL_ANSWER, originate a new call.

Now, How can I get the uuid of the new call and receive events for this new
call.? I want to receive the events for both uuids.



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[Freeswitch-users] Re corded file as voicemail.

2009-10-05 Thread Nagalenoj

Is it possible to treat a recorded voice as voice mail?

Assume that, I've recorded a conversation and I want this recorded file to
be treated like voicemail. So, I could check it like voicemail!!

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[Freeswitch-users] Listening to a connected call [barge in]

2009-10-01 Thread Nagalenoj

In ES outbound, I need to do the following, 
* A calls 2000(FS ES outbound extension)
* In the script, It'll answer the call, play some files and get the reply
from A(as voice).
* Simultaneously(when doing the above), the script has to call B.
* When B attends the call, B has to listen to the live conversation between
2000 & A.

How should I do.?

I've tried this with async mode and by listening to the events. But I
couldn't do it. Help me to do this..

Regards,
Nagalenoj
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Re: [Freeswitch-users] Mod_perl or ESL

2009-09-22 Thread Nagalenoj

I need to handle some hundreds of call. So, which one can I opt?


Nagalenoj wrote:
> 
> Dear friends,
> I want to know which is the better way to do route calls and control
> calls. I've did a experiment which can be done in both ways, Mod_perl and
> ESL. I don't know which one is better to take.
> When I see some earlier posts, It is given like Mod_perl has some
> limitations and I don't know what kind of limitations they are.,
> Can someone say which is better to use and how it is better?
> 

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[Freeswitch-users] Mod_perl or ESL

2009-09-19 Thread Nagalenoj

Dear friends,
I want to know which is the better way to do route calls and control
calls. I've did a experiment which can be done in both ways, Mod_perl and
ESL. I don't know which one is better to take.
When I see some earlier posts, It is given like Mod_perl has some
limitations and I don't know what kind of limitations they are.,
Can someone say which is better to use and how it is better?
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Re: [Freeswitch-users] ESL: DTMF event is not coming

2009-09-08 Thread Nagalenoj

Yes.!! I missed "async" keyword in dialplan in freeswitch-1.0.4. Thanks..


Anthony Minessale-2 wrote:
> 
> did you specify the "async" keyword to the socket app in your dialplan?
> 
> On Mon, Sep 7, 2009 at 2:47 AM, Nagalenoj  wrote:
> 
>>
>> Dear friends,
>>   I am using freeswitch-1.0.4. When I execute the sample
>> script(/libs/esl/perl/server2.pl), it is not receiving the DTMF events.
>> When
>> I execute the same program in freeswitch-1.0.3, it's receiving the event.
>> Do
>> I miss something to configure/upgrade.?
>> --
>> View this message in context:
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>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
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> 
> 
> 
> -- 
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[Freeswitch-users] ESL: DTMF event is not coming

2009-09-07 Thread Nagalenoj

Dear friends,
   I am using freeswitch-1.0.4. When I execute the sample
script(/libs/esl/perl/server2.pl), it is not receiving the DTMF events. When
I execute the same program in freeswitch-1.0.3, it's receiving the event. Do
I miss something to configure/upgrade.?
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Re: [Freeswitch-users] How can it be done with freeswitch?

2009-08-28 Thread Nagalenoj

Thanks for your help. Is it the only possible way.?
Else, will it be done with the functions listed in the below url,
http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes

I was trying to do with these classes and functions.


mercutioviz wrote:
> 
> Nagalenoj,
> 
> This is definitely possible to do with FreeSWITCH. However, it is not
> exactly simple. You have some learning to do. My recommendation is to
> learn
> about event socket. Most likely you'll need to use outbound event socket.
> (Read about event socket and ESL, the event socket library, on the wiki.)
> 
> I also recommend that you look at some of the example scripts in
> src/libs/esl/perl/ which show some of the ways to handle calls with a
> script. You've already done a good job of breaking down the steps, now
> experiment with handling each of these steps with a script. We don't have
> extensive Perl script examples which means we would love to see your code
> when you have it all figured out! :)
> 
> -MC
> 
> On Thu, Aug 27, 2009 at 11:44 PM, Nagalenoj  wrote:
> 
>>
>> Dear friends,
>>I am using freeswitch for the last one month and trying to write perl
>> scripts to do some tasks.
>> I am trying a new thing, which is something that I don't how to do it and
>> I
>> don't whether it can be done with freeswitch.   my requirement is as
>> follows,
>> A calls B[This extension has to execute a script].
>> The script has to do the following,
>>  * Cut the call and callback to the number.
>>  * Get a phone number[DTMF] from A.
>>  * Dial the number(C) and bridge A and C.
>>  * When C cuts the call, the control has to comeback to
>> script.
>>  * Ask another number to A to dial out, get another phone
>> number[DTMF] from A.
>>  * Dial the number(D) and bridge A and D.
>>
>> Please, help me to do this. Is there any other documentation to read
>> about
>> the perl freeswitch functions.
>> --
>> View this message in context:
>> http://www.nabble.com/How-can-it-be-done-with-freeswitch--tp25184845p25184845.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
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[Freeswitch-users] How can it be done with freeswitch?

2009-08-28 Thread Nagalenoj

Dear friends,
I am using freeswitch for the last one month and trying to write perl
scripts to do some tasks. 
I am trying a new thing, which is something that I don't how to do it and I
don't whether it can be done with freeswitch.   my requirement is as
follows,
 A calls B[This extension has to execute a script].
 The script has to do the following,
  * Cut the call and callback to the number.
  * Get a phone number[DTMF] from A.
  * Dial the number(C) and bridge A and C.   
  * When C cuts the call, the control has to comeback to script.
  * Ask another number to A to dial out, get another phone
number[DTMF] from A.
  * Dial the number(D) and bridge A and D.

Please, help me to do this. Is there any other documentation to read about
the perl freeswitch functions.
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