[Freeswitch-users] Auto dialing ...

[Anthony Minessale]

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 at gmail.com> wrote:

> 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 at 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 at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090123/290f76a7/attachment-0002.html