Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[Freeswitch-users] Newbie Questions


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users
View previous topic :: View next topic  
Author Message
vitolini at gmail.com
Guest





PostPosted: Fri Oct 03, 2008 8:48 am    Post subject: [Freeswitch-users] Newbie Questions Reply with quote

Hi All,

I am familiar with Asterisk and doing some testing for my next project. I have had some difficulties Asterisk, and now researching FreeSwitch hoping that it has some out-of-the box answers for my questions.

Basically I want to implement a Click 2 Call service. Very simple, user types in his/her number on a website, that number is called, after it's answered, the company number is called and they are bridged together.

1) Is there a way to communicate with FreeSwitch programatically and issue commands such as initiate calls etc ? (ver much like manager API in Asterisk)

2) If you initiate a call from the software and then once its answered call a 2nd number. How do you bridge them?

3) After the 2 numbers talk and hang up. How does your cdr look like? Do you have 2 cdr's that correspond to both calls or just 1 after both numbers are bridged together? This is one of the problems I can't solve with Asterisk as it generates only 1 cdr after the 2 calss are bridged. The reason for this request is, in case of a Click2Call service, you are charged for both calls by your SIP provider therefore you need to be able to track both calls for invoices/payments etc.

4) Is there a way to programatically know if a call has been asnwered or not and act based upon that. I understand the cdr contains that information. But what I want is, if the call is not answered maybe I can play a prerecorded message or take them to the voicemail or whatnot. So I need a way to do a flow-control based on if the call has been asnwered or not in the dialplan. Does that exist? If so can you point me to some resources?

Thanks,

Vito A.
Back to top
mike at jerris.com
Guest





PostPosted: Fri Oct 03, 2008 9:09 am    Post subject: [Freeswitch-users] Newbie Questions Reply with quote

On Oct 3, 2008, at 4:54 AM, Vito Andolini wrote:
Quote:
Hi All,

I am familiar with Asterisk and doing some testing for my next project. I have had some difficulties Asterisk, and now researching FreeSwitch hoping that it has some out-of-the box answers for my questions.

Basically I want to implement a Click 2 Call service. Very simple, user types in his/her number on a website, that number is called, after it's answered, the company number is called and they are bridged together.

1) Is there a way to communicate with FreeSwitch programatically and issue commands such as initiate calls etc ? (ver much like manager API in Asterisk)



There is an interface that we call the fsapi interface that can be accessed in many ways, including over a socket method similar to a combination of AMI and FAGI:


http://wiki.freeswitch.org/wiki/Event_Socket


and xmlrpc:


http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC

Quote:
2) If you initiate a call from the software and then once its answered call a 2nd number. How do you bridge them?



You can do this all in one command:


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


Quote:
3) After the 2 numbers talk and hang up. How does your cdr look like? Do you have 2 cdr's that correspond to both calls or just 1 after both numbers are bridged together? This is one of the problems I can't solve with Asterisk as it generates only 1 cdr after the 2 calss are bridged. The reason for this request is, in case of a Click2Call service, you are charged for both calls by your SIP provider therefore you need to be able to track both calls for invoices/payments etc.



We can do either per leg or combined cdr's


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


We have multiple supported formats for cdr:


http://wiki.freeswitch.org/wiki/Mod_xml_cdr
http://wiki.freeswitch.org/wiki/Mod_cdr_csv

Quote:
4) Is there a way to programatically know if a call has been asnwered or not and act based upon that. I understand the cdr contains that information. But what I want is, if the call is not answered maybe I can play a prerecorded message or take them to the voicemail or whatnot. So I need a way to do a flow-control based on if the call has been asnwered or not in the dialplan. Does that exist? If so can you point me to some resources?



There are several approaches you could take to this. You could do this all in dialplan if there is not any real forking other than if the call worked. You can use the variables:


http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail


and just playing the sounds in the dial-plan after a bridge line.




Mike
Back to top
vitolini at gmail.com
Guest





PostPosted: Fri Oct 03, 2008 2:16 pm    Post subject: [Freeswitch-users] Newbie Questions Reply with quote

Michael,

Thanks a lot for the answers, it answers my questions, I guess the next thing would be downloading and trying FS Smile

Also, as I saw in the wiki, there is no built in support for mysql right? The solution that was suggested was setting up a cron to import the csv files into mysql, is this still what's available or are there any updates on this?

Thank,

Vito A.

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Friday, October 03, 2008 7:06 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Newbie Questions




On Oct 3, 2008, at 4:54 AM, Vito Andolini wrote:
Quote:
Hi All,

I am familiar with Asterisk and doing some testing for my next project. I have had some difficulties Asterisk, and now researching FreeSwitch hoping that it has some out-of-the box answers for my questions.

Basically I want to implement a Click 2 Call service. Very simple, user types in his/her number on a website, that number is called, after it's answered, the company number is called and they are bridged together.

1) Is there a way to communicate with FreeSwitch programatically and issue commands such as initiate calls etc ? (ver much like manager API in Asterisk)



There is an interface that we call the fsapi interface that can be accessed in many ways, including over a socket method similar to a combination of AMI and FAGI:


http://wiki.freeswitch.org/wiki/Event_Socket


and xmlrpc:


http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC

Quote:
2) If you initiate a call from the software and then once its answered call a 2nd number. How do you bridge them?



You can do this all in one command:


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


Quote:
3) After the 2 numbers talk and hang up. How does your cdr look like? Do you have 2 cdr's that correspond to both calls or just 1 after both numbers are bridged together? This is one of the problems I can't solve with Asterisk as it generates only 1 cdr after the 2 calss are bridged. The reason for this request is, in case of a Click2Call service, you are charged for both calls by your SIP provider therefore you need to be able to track both calls for invoices/payments etc.



We can do either per leg or combined cdr's


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


We have multiple supported formats for cdr:


http://wiki.freeswitch.org/wiki/Mod_xml_cdr
http://wiki.freeswitch.org/wiki/Mod_cdr_csv

Quote:
4) Is there a way to programatically know if a call has been asnwered or not and act based upon that. I understand the cdr contains that information. But what I want is, if the call is not answered maybe I can play a prerecorded message or take them to the voicemail or whatnot. So I need a way to do a flow-control based on if the call has been asnwered or not in the dialplan. Does that exist? If so can you point me to some resources?



There are several approaches you could take to this. You could do this all in dialplan if there is not any real forking other than if the call worked. You can use the variables:


http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail


and just playing the sounds in the dial-plan after a bridge line.




Mike

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.173 / Virus Database: 270.7.5/1702 - Release Date: 10/2/2008 9:35 PM
Back to top
thomas.mangin at exa-n...
Guest





PostPosted: Sat Oct 04, 2008 2:29 am    Post subject: [Freeswitch-users] Newbie Questions Reply with quote

Another way would be to export to a Radius server using a MySQL backend.

I am not using the feature so I can not report how well it works but I
am confident that "it should just work" and that if it does not, it will
be fixed Smile

Thomas

Vito Andolini wrote:
Quote:
Michael,

Thanks a lot for the answers, it answers my questions, I guess the next
thing would be downloading and trying FS Smile

Also, as I saw in the wiki, there is no built in support for mysql
right? The solution that was suggested was setting up a cron to import
the csv files into mysql, is this still what's available or are there
any updates on this?

Thank,

Vito A.

------------------------------------------------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of
*Michael Jerris
*Sent:* Friday, October 03, 2008 7:06 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Newbie Questions


On Oct 3, 2008, at 4:54 AM, Vito Andolini wrote:

Quote:
Hi All,

I am familiar with Asterisk and doing some testing for my next
project. I have had some difficulties Asterisk, and now researching
FreeSwitch hoping that it has some out-of-the box answers for my
questions.

Basically I want to implement a Click 2 Call service. Very simple,
user types in his/her number on a website, that number is called,
after it's answered, the company number is called and they are bridged
together.

1) Is there a way to communicate with FreeSwitch programatically and
issue commands such as initiate calls etc ? (ver much like manager API
in Asterisk)

There is an interface that we call the fsapi interface that can be
accessed in many ways, including over a socket method similar to a
combination of AMI and FAGI:

http://wiki.freeswitch.org/wiki/Event_Socket

and xmlrpc:

http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC

Quote:
2) If you initiate a call from the software and then once its answered
call a 2nd number. How do you bridge them?

You can do this all in one command:

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

Quote:
3) After the 2 numbers talk and hang up. How does your cdr look like?
Do you have 2 cdr's that correspond to both calls or just 1 after both
numbers are bridged together? This is one of the problems I can't
solve with Asterisk as it generates only 1 cdr after the 2 calss are
bridged. The reason for this request is, in case of a Click2Call
service, you are charged for both calls by your SIP provider therefore
you need to be able to track both calls for invoices/payments etc.

We can do either per leg or combined cdr's

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

We have multiple supported formats for cdr:

http://wiki.freeswitch.org/wiki/Mod_xml_cdr
http://wiki.freeswitch.org/wiki/Mod_cdr_csv

Quote:
4) Is there a way to programatically know if a call has been asnwered
or not and act based upon that. I understand the cdr contains that
information. But what I want is, if the call is not answered maybe I
can play a prerecorded message or take them to the voicemail or
whatnot. So I need a way to do a flow-control based on if the call has
been asnwered or not in the dialplan. Does that exist? If so can you
point me to some resources?

There are several approaches you could take to this. You could do this
all in dialplan if there is not any real forking other than if the call
worked. You can use the variables:

http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail

and just playing the sounds in the dial-plan after a bridge line.


Mike

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.173 / Virus Database: 270.7.5/1702 - Release Date:
10/2/2008 9:35 PM


------------------------------------------------------------------------

_______________________________________________
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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services