VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
View previous topic :: View next topic |
Author |
Message |
sicfslist at gmail.com Guest
|
Posted: Fri Jan 23, 2009 5:08 pm Post subject: [Freeswitch-users] Auto dialing ... |
|
|
OK ... Here goes another I'm doing this with AST ... but I want to move it to FS. Searched via google site:lists.freeswitch.org auto dialer and others ... nothing useful.
Today I have a platform for auto dialing with AST (centrally managed ... about 10 machines) and we do this:
-- Remote machines query central DB for numbers to call based on certain configs
-- Use AMI to generate the call
-- If call gets answered, extension info queried via rta (central db again)
The nice thing about all of this is it's relatively easy to manage (through one central web interface we built) and it works ... the bad part is reporting ... as anyone knows on this list that has used AST for auto dialing in this way (via .call or AMI) every call looks like it fails instead of showing a real cause code.
So ... conceptually I'm trying to accomplish the same thing ...
Today we use FS a lot for termination of VoIP traffic ... all done via XML_CURL ... which is awesome!
Would like to do something like:
-- originate request
-- on answer XML_CURL posts info |
|
Back to top |
|
|
egghunt at gmail.com Guest
|
Posted: Fri Jan 23, 2009 5:40 pm Post subject: [Freeswitch-users] Auto dialing ... |
|
|
On Fri, Jan 23, 2009 at 8:04 PM, Shelby Ramsey <sicfslist@gmail.com (sicfslist@gmail.com)> wrote:
Quote: | OK ... Here goes another I'm doing this with AST ... but I want to move it to FS. Searched via google site:lists.freeswitch.org auto dialer and others ... nothing useful.
Today I have a platform for auto dialing with AST (centrally managed ... about 10 machines) and we do this:
-- Remote machines query central DB for numbers to call based on certain configs
-- Use AMI to generate the call
-- If call gets answered, extension info queried via rta (central db again)
The nice thing about all of this is it's relatively easy to manage (through one central web interface we built) and it works ... the bad part is reporting ... as anyone knows on this list that has used AST for auto dialing in this way (via .call or AMI) every call looks like it fails instead of showing a real cause code.
So ... conceptually I'm trying to accomplish the same thing ...
Today we use FS a lot for termination of VoIP traffic ... all done via XML_CURL ... which is awesome!
Would like to do something like:
-- originate request
-- on answer XML_CURL posts info
|
Auto dialing is one of the many areas where freeswitch is much superior than asterisk. You can accomplish what you need in some ways, one would be to listen for a CHANNEL_ANSWER event on event_socket interface and, then, take whatever needed action.
--
Arnaldo M Pereira
http://lustyscripps.wordpress.com |
|
Back to top |
|
|
brian at freeswitch.org Guest
|
Posted: Fri Jan 23, 2009 5:42 pm Post subject: [Freeswitch-users] Auto dialing ... |
|
|
http://wiki.freeswitch.org/wiki/Mod_commands#originate
/b
On Jan 23, 2009, at 4:33 PM, Arnaldo de Moraes Pereira wrote:
Quote: | On Fri, Jan 23, 2009 at 8:04 PM, Shelby Ramsey <sicfslist@gmail.com (sicfslist@gmail.com)> wrote:
Quote: | OK ... Here goes another I'm doing this with AST ... but I want to move it to FS. Searched via google site:lists.freeswitch.org auto dialer and others ... nothing useful.
Today I have a platform for auto dialing with AST (centrally managed ... about 10 machines) and we do this:
-- Remote machines query central DB for numbers to call based on certain configs
-- Use AMI to generate the call
-- If call gets answered, extension info queried via rta (central db again)
The nice thing about all of this is it's relatively easy to manage (through one central web interface we built) and it works ... the bad part is reporting ... as anyone knows on this list that has used AST for auto dialing in this way (via .call or AMI) every call looks like it fails instead of showing a real cause code.
So ... conceptually I'm trying to accomplish the same thing ...
Today we use FS a lot for termination of VoIP traffic ... all done via XML_CURL ... which is awesome!
Would like to do something like:
-- originate request
-- on answer XML_CURL posts info
|
Auto dialing is one of the many areas where freeswitch is much superior than asterisk. You can accomplish what you need in some ways, one would be to listen for a CHANNEL_ANSWER event on event_socket interface and, then, take whatever needed action.
--
Arnaldo M Pereira
http://lustyscripps.wordpress.com
_______________________________________________
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 |
|
|
anthony.minessale at g... Guest
|
Posted: Fri Jan 23, 2009 5:44 pm Post subject: [Freeswitch-users] Auto dialing ... |
|
|
Does AST mean Asterisk Open Source PBX ?
If so, then yes I am familiar with it's archetechure as I am a former developer from that project.
You have 3 choices with FreeSWITCH
1) You can open a dedicated connection to mod_event_socket or XMLRPC per call and issue the originate command from there:
This will block until you know for sure the outcome of the attempt. If it's success it will give you the uuid if not it gives you the cause code.
2) You can use a single mod_event_socket or XMLRPC connection to send all calls but use the bgapi mechanism which will do the same as above
only asynchronously, The command will return immediately and the result will be fired as an event that you can pick up on the same or different event_socket connection or
other event consumer such as a custom C,perl,lua etc module.
3) You can use mod_xml_cdr to generate detailed 1-file-per-leg call files that will tell you when where and why the calls failed or did not fail.
On Fri, Jan 23, 2009 at 4:04 PM, Shelby Ramsey <sicfslist@gmail.com (sicfslist@gmail.com)> wrote:
Quote: | OK ... Here goes another I'm doing this with AST ... but I want to move it to FS. Searched via google site:lists.freeswitch.org auto dialer and others ... nothing useful.
Today I have a platform for auto dialing with AST (centrally managed ... about 10 machines) and we do this:
-- Remote machines query central DB for numbers to call based on certain configs
-- Use AMI to generate the call
-- If call gets answered, extension info queried via rta (central db again)
The nice thing about all of this is it's relatively easy to manage (through one central web interface we built) and it works ... the bad part is reporting ... as anyone knows on this list that has used AST for auto dialing in this way (via .call or AMI) every call looks like it fails instead of showing a real cause code.
So ... conceptually I'm trying to accomplish the same thing ...
Today we use FS a lot for termination of VoIP traffic ... all done via XML_CURL ... which is awesome!
Would like to do something like:
-- originate request
-- on answer XML_CURL posts info
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400 |
|
Back to top |
|
|
|
|
|
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
|