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
On Fri, Jan 23, 2009 at 8:04 PM, Shelby Ramsey <sicfslist@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
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.
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-- Arnaldo M Pereira http://lustyscripps.wordpress.com
http://wiki.freeswitch.org/wiki/Mod_commands#originate /b On Jan 23, 2009, at 4:33 PM, Arnaldo de Moraes Pereira wrote:
On Fri, Jan 23, 2009 at 8:04 PM, Shelby Ramsey <sicfslist@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
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.
_______________________________________________ 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
-- 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
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> 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
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-- 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
Hello - I am brand-new to VOIP and have set up freeswitch on our pFsense firewall. It works great with X-lite. I would like to hook up a linksys SPA941 to the system, and am having trouble finding any documentation on how to configure it. Where do I start? Thanks - Mark
Mark, Welcome to the FreeSWITCH Community. Check this out http://voxilla.com/tools/device-configuration-wizard/linksys-spa941-configur... Also when you post to the list please click new message, input the freeswitch-users@lists.freeswitch.org and do a new subject and click send. When you reply to an existing email, change the subject and contents you hijack a thread which can cause your email to go un-noticed which could result in your question not being answered sometimes. ;) Anyway the config wizard would help you get started. Thanks, Brian On Jan 23, 2009, at 4:47 PM, Mark wrote:
Hello -
I am brand-new to VOIP and have set up freeswitch on our pFsense firewall. It works great with X-lite. I would like to hook up a linksys SPA941 to the system, and am having trouble finding any documentation on how to configure it. Where do I start?
Thanks -
Mark
A very simple (maybe) question. I have freeswitch installed behind a firewall with needs ports open I have users internal (connected to local lan address) and users external (connected to my public internet IP address) All users are configured in the directory/default. When I call from a lan user a external user all work fine When I call from a external user a lan user the call drop after 30-32 seconds When I call from a external user other external user the call drop after 30-32 seconds When I call from a lan user other lan user all work fine I thing is a NAT problem but I don't know how solve it Regards - Andrea -
participants (6)
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Andrea
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Anthony Minessale
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Arnaldo de Moraes Pereira
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Brian West
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Mark
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Shelby Ramsey