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[nik.middleton at noble...]

Hi Guys,

As a long time Asterisk user, I’m looking into freeswitch as an alternative mainly due to (list multiple reasons here)

Can anyone give me a pointer as to how I would achieve the following?

I need to replicate an emergency broadcast system currently running under Asterisk.

At the moment, I run through a Mysql database and using the manager API, issues an Originate command to dial a number.

When the call is answered, a message is played, and the recipient has the option of hitting a digit to confirm receipt. I then call an AGI script to update the database.

Is this fairly easy to do in Freeswitch?

Not looking for code, just some pointers as to what’s available to do the above /

Regards,

[msc at freeswitch.org]

Nik,


Welcome to FreeSWITCH! The short answer is "yes, FS can do that." The first thing that you should do is unlearn "the Asterisk way" of thinking. Usually there is an elegant way of doing things in FS that wasn't possible in Ast.


I would recommend that you start by looking at the event socket, which is somewhat analogous to the AMI only cooler. I have personally done something similar to this using the event socket and a Perl script. The key is to learn the syntax of the originate command. (definitely hit the wiki and IRC channel)


Are you using TDM cards for this? Just curious.


-MC (IRC nick: mercutioviz)

Sent from my iPhone

On Feb 2, 2009, at 3:35 PM, "Nik Middleton" <nik.middleton@noblesolutions.co.uk (nik.middleton@noblesolutions.co.uk)> wrote:

[nik.middleton at noble...]

Thanks for that, coming from a C++ background it’s a refreshing change to be looking at something that seems logical and efficient.

I’d briefly looked at the event socket and wondered if that was the way to go. I presume that there’s some sort of event generation that can trigger and external process as well somewhere, though all I need to do is update mysql (hopefully using some sort of pooled connection)

I’m not using a TDM card, I have a direct interconnect with the PSTN breakout provider with 1,500 channels available to me. I’m finding Asterisk proving to be less than stable at high call volumes and load values spike at more than 100 calls with billing/accounting in place, hence my interest in FS. The only thing that’s concerning me is XML at the moment. Lots of code and very wordy. I’m sure I’ll appreciate why XML given time

Regards,


From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael S Collins
Sent: 03 February 2009 01:17
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Generating calls from external source


Nik,



Welcome to FreeSWITCH! The short answer is "yes, FS can do that." The first thing that you should do is unlearn "the Asterisk way" of thinking. Usually there is an elegant way of doing things in FS that wasn't possible in Ast.



I would recommend that you start by looking at the event socket, which is somewhat analogous to the AMI only cooler. I have personally done something similar to this using the event socket and a Perl script. The key is to learn the syntax of the originate command. (definitely hit the wiki and IRC channel)

Are you using TDM cards for this? Just curious.



-MC (IRC nick: mercutioviz)


Sent from my iPhone


On Feb 2, 2009, at 3:35 PM, "Nik Middleton" <nik.middleton@noblesolutions.co.uk (nik.middleton@noblesolutions.co.uk)> wrote:

[dave at 3c.co.uk]

Hi Nik,

Here's a snipped in Perl that launches an outbound call:


if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => '127.0.0.1', PeerPort => 8021)) {
print $sock "auth XXX\n\n";
print $sock "api originate {softivr_id=$siid,src_softivr_id=$siid,softivr_outdial=true}sofia/frombt/$ntd@1.2.3.4 ([email]softivr_outdial=true}sofia/frombt/@1.2.3.4[/email]) $service\n\n";
$sock->close();
}


- it does no error checking or anything, but (line by line) it:
- opens a socket to the event socket interface
- authenticates
- issues an originate which dials out to the number in $ntd. The bits in {} set a bunch of variables on the channel, which are used by the software which processes the call later on. The call is linked to the extension in $service - FS looks this up in the dialplan - which handles our end.
- closes the socket


Cheers --


Dave

[raul at etellicom.com]

In addition do David's suggestion, you probably want to have your
application to watch for some specific events after the call is
originated and take action based on them. For example, you could watch
for the CHANNEL_ANSWER event and play some audio file waiting for some
digit, which is generated by the DTMF event.
To watch only for those specific events, you should do the following
just after authentication (still using Perl as an example, but the
mod_event_socket is language agnostic), then you will receive those
events from FreeSWITCH through the socket stream:

...
print $sock "auth XXX\n\n";
print $sock "event plain CHANNEL_ANSWER DTMF\n\n";
...

To see a list of available events, please look at the following wiki
pages:
http://wiki.freeswitch.org/wiki/Mod_event_socket#event
http://wiki.freeswitch.org/wiki/Event_list

Regards,

Raul

On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote:

[anthony.minessale at g...]

There is also an event socket library written in C called esl that is in the fs tree in the libs directory.
This has the ability to establish connections both inbound and outbound from FS.

There is also a perl module FreeSWITCH::Client that mr collins may be interested in in the tree as well.


On Tue, Feb 3, 2009 at 7:12 AM, Raul Fragoso <raul@etellicom.com (raul@etellicom.com)> wrote:

[msc at freeswitch.org]

On Tue, Feb 3, 2009 at 6:09 AM, Anthony Minessale
<anthony.minessale@gmail.com> wrote:

[nik.middleton at noble...]

Are you suggesting that I should process the call externally instead of
using the dialplan? That would be neat as the audio file select could
be driven from the db select for the number. I presume that I could
also bridge the call to another number as well dependant on DTMF
selection?

Regards


-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Raul
Fragoso
Sent: 03 February 2009 13:12
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Generating calls from external source

In addition do David's suggestion, you probably want to have your
application to watch for some specific events after the call is
originated and take action based on them. For example, you could watch
for the CHANNEL_ANSWER event and play some audio file waiting for some
digit, which is generated by the DTMF event.
To watch only for those specific events, you should do the following
just after authentication (still using Perl as an example, but the
mod_event_socket is language agnostic), then you will receive those
events from FreeSWITCH through the socket stream:

...
print $sock "auth XXX\n\n";
print $sock "event plain CHANNEL_ANSWER DTMF\n\n";
...

To see a list of available events, please look at the following wiki
pages:
http://wiki.freeswitch.org/wiki/Mod_event_socket#event
http://wiki.freeswitch.org/wiki/Event_list

Regards,

Raul

On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote:

[sicfslist at gmail.com]

Nik,
There are a lot of ways to make FS dial out and deliver messaging etc. We are going through the process of replacing * for this purpose. For us (getting started with the help of our friends here on the list) it has been pretty easy.

With * we were using AMI to originate calls ... to migrate to FS we just changed that to use event_socket with bgapi to originate the call and connect the call to a context and extension. There are several ways to get the dialplan to FS after that ... a script, xml_curl, or statically configured in the conf directory.

So as an example the application we have just logs into the FS socket (similar to * but much better) and then rips off calls like this:

bgapi originate{$set_some_vars}sofia/external/$ANI@$IP:$PORT $EXTENSION xml $CONTEXT

The beauty of it all is that:
-- a lot of flexibility in what you can do (like drive the call through events)
-- the CDR reporting is about 3 million times better than *
-- obviously higher capacity

I'd start playing with event_socket and some static dialplans to get the feel for it ... but if you have an application written already to work with * (i.e. the logic and backend) it will be very easy to migrate and you'll be glad you did it!

Shelby



On Tue, Feb 3, 2009 at 10:53 AM, Nik Middleton <nik.middleton@noblesolutions.co.uk (nik.middleton@noblesolutions.co.uk)> wrote:

[msc at freeswitch.org]

On Tue, Feb 3, 2009 at 8:53 AM, Nik Middleton
<nik.middleton@noblesolutions.co.uk> wrote:

[nicolas at medularis.com]

Nik,

There's also a PHP library fs_sock.php under contrib in the source
code. I used it to create a simple app that originates calls and then
run some other sutff when it detects the call has ended. The actual
call originate command is executed inside a javascript file which is
run using bgapi jsrun. The js script also makes a POST request to an
external URL using CURL.

There's plenty to play around with, Freeswitch is really great, and
mostly easy, a world of difference with *.

Good luck!

Nicolas

On Tue, Feb 3, 2009 at 2:13 PM, Shelby Ramsey <sicfslist@gmail.com> wrote:

[raul at etellicom.com]

Hi Nik,

That's one possibility, yes. You could use mod_xml_curl to provide the
dial-plan on the fly and then use mod_event_socket to send commands to
FS and process events. That's exactly what I do actually, we have an IVR
engine that is driven by mod_event_socket and another module that
provides the XML dial-plan through mod_xml_curl.
The beauty of FS is that you have many options to tack a problem, and
all of those options are very elegant. I suggest looking at
mod_event_socket first and then decide if you can live with the static
dial-plan or go to a more dynamic dial-plan via mod_xml_curl.

Regards,

Raul

On Tue, 2009-02-03 at 16:53 +0000, Nik Middleton wrote: