Generating calls from external source
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,
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
wrote:
Hi Guys,
As a long time Asterisk user, I’m looking into freeswitch as an alte rnative 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 d o the above /
Regards,
_______________________________________________ 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 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> wrote: 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, _______________________________________________ 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://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 $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
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
wrote: 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, _______________________________________________ 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
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:
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 $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
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> wrote:
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, _______________________________________________ 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
_______________________________________________ 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
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> wrote:
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:
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 $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
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> wrote:
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, _______________________________________________ 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
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
On Tue, Feb 3, 2009 at 6:09 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
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.
As a matter of fact that is the module I used for my outbound IVR application. It simply handled the communications between my perl script and my FS instance. The script would read in pre-formatted originate strings from a text file that had been previously generated by another application. Then all I had to do was specify how many concurrent channels that I wanted - kind of like a throttle - and then I let the script go. I used the "bgapi originate" syntax so that I wouldn't have to wait to see what happened with each origination attempt. Then about every second or so I would issue an "oz dump 1" and parse the results to count how many b channels were in use. If the number of b channels in use was >= my throttle limit then I'd pause the script for 1000ms and then issue the oz dump again until the number of b channels in use had dropped down below my limit. Nothing too fancy. You're welcome to review my script, originate syntax, and dialplan entries if you are interested. -MC
On Tue, Feb 3, 2009 at 7:12 AM, Raul Fragoso <raul@etellicom.com> wrote:
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:
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 $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
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> wrote:
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, _______________________________________________ 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
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
_______________________________________________ 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
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:
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 $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
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> wrote:
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, _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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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> wrote:
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:
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 $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
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> wrote:
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, _______________________________________________ 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
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
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_______________________________________________ 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
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:
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> wrote:
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:
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 $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
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> wrote:
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, _______________________________________________ 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
Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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On Tue, Feb 3, 2009 at 8:53 AM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote:
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
I'm not saying you should, merely that you could. What I did was create a bunch of extensions in my dialplan that handled various steps of the IVR outbound call: start, answered, busy, not answered, SIT tones, etc. So my originate command would originate the call (A leg) and drop the B leg into the dialplan at the "start" extension and then it goes from there. It listens for early media busy or SIT tones and also does an "execute_on_answer" to the extension that does the actual IVR. (Only need the IVR on an answered call.) If the call is not answered after 25 seconds then I run a Lua script that checks for the presence of certain channel variables that I set with the "tone_detect" application (busy and SIT). If none of those are present then I assume the call went unanswered and do the post-processing.
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?
Yes, you can do this as well. You can build an IVR in XML or you can build in a scripting language like Lua: demo IVR: http://svn.freeswitch.org/svn/freeswitch/trunk/conf/autoload_configs/ivr.con... Lua IVR info: http://wiki.freeswitch.org/wiki/IVR#Lua_IVRs Sorry if this is all a bit overwhelming, but you'll be glad that you dove in to FS because it does soooo much and does it so well. Enjoy! -MC
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:
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 $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
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> wrote:
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, _______________________________________________ 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
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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:
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:
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 $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
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> wrote:
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 /
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participants (8)
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Anthony Minessale
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David Knell
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Michael Collins
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Michael S Collins
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Nicolas Brenner
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Nik Middleton
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Raul Fragoso
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Shelby Ramsey