I see your dilemma. To keep things dynamic you definitely want to use your XML_CURL stuff. Like you said, nothing static. Can you post a sample call-flow in plain English? I'm curious about something. I don't want to say anything else. Just post a simple call flow: Initiate call Wait for answer/busy/timeout/invalid On busy/timeout/invalid: update db and move on On answer play greeting, accept digit, route based on digit Something like that would help me conceptualize what you are trying to do. Thanks MC P.S. - I've done a little bit of outbound IVR calling so hopefully I can assist you. On Fri, Jan 23, 2009 at 3:39 PM, Shelby Ramsey <sicfslist@gmail.com> wrote:
Anthony / Michael, Thanks for the quick responses. What I don't want to do is "drive the call" (by that listen on a socket ... do this on this event ... or anything else that my very limited FS foo would break) ... Just want to start it and then give it instructions on where to go. So I guess a better question would be ... how do I give directions to FS for this (and I get the 1st part ... that's obvious ... really lost on the DTMF digit part) ... and please keep in mind we're talking hundreds of extensions / IVR's and distributed machines so I can't have any dependancy on static conf files other than maybe something like what Michael mentioned where I point every call to something: [campaign] exten => 100,1,ANSWER() exten => 100,n,PLAYBACK(somefile) exten => 100,n,BACKGROUND(somefile) exten => 100,n,WAITEXTEN(4) exten => 100,n,HANGUP() but in that same context is someone triggers DTMF: exten => 1,1,DOSOMETHING exten => 2,1,DOSOMETHING I was imaging issuing originate via XML_RPC ... something like originate sofia/$ANI@$IP $SOMEEXTEN then on answer when FS tries to connect to $SOMEEXTEN it will ask me what to do via xml_curl ... where I would normally respond with something like this: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="FS RESPONSE"> <context name="public"> <extension name="$EXTEN"> <condition field="destination_number" expression=""> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="call_timeout=180"/> <action application="set" data="proxy_media=true"/> <action application="set" data="pass_rfc2833=true"/> <action application="set" data="accountcode=$CUSTOMER" /> <action application="set" data="origination_caller_id_name=NULL" /> <action application="set" data="origination_caller_id_number=$CIDNUM" /> <action application="set" data="effective_caller_id_name=NULL" /> <action application="set" data="effective_caller_id_number=$CIDNUM" /> <action application="set" data="userfield=$BUNCHOFCRAPFORMYCDR" /> <action application="bridge" data="sofia/external/$ANI@$PROVIDERIP" /> </condition> </extension> </context> </section> </document> The challenge I've got is I have no idea how to do stuff like the IVR mentioned above (the playback part is easy) ... but I can't grasp conceptually how to get the "context" with "multiple extensions" part back to FS via this method (is it possible?)... Sorry for what is probably a very simple answer and any AST references (but I've been using it in heavy production environments for about 5 years). Just trying to "port" what I do today without making my brain melt out of my ears (and it doesn't take much for that to happen). Shelby PS ... Really enjoy the list. I usually fall out of my chair laughing once a day from your remarks Anthony. Keep it coming!
On Fri, Jan 23, 2009 at 4:55 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
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 4:39 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Jan 23, 2009 at 2:15 PM, Shelby Ramsey <sicfslist@gmail.com> wrote:
Sorry for the double post ... actually hit send too early ... 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 ... 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 (not to xml cdr ... and the "proxying" of media) ... Would like to do something like: -- originate request (looks simple enough) -- on answer XML_CURL posts info
Several choices, depending upon how much you want it handled inside the dialplan vs. handled in the scripting language. For the sake of testing you could do something like this: <extension name="ivr-start"> <condition field="destination_number" expression="ivr_whatever"> <action application="set" data="execute_on_answer=transfer IVR_ANSWER XML default"/> <!-- rest of dialplan --> </condition> </extension>
Then have: <extension name="ivr-answer"> <condition field="destination_number" expression="IVR_ANSWER"> <action application="lua" data="post-info.lua ${some_important_value}"/> </condition> </extension>
This would have any answered call go to the "ivr-answer" extension while unanswered calls could stay in the ivr-start extension to get properly handled. (Busy, no answer, invalid/SIT, etc.)
You could then have the "ivr-answer" extension do whatever is appropriate, like listen for digits, play announcement, beg for money, etc. :)
-MC
But for the life of me I can't figure out how to translate this into the xml response ... [campaign] exten => 100,1,ANSWER() exten => 100,n,WAIT(2) exten => 100,n,BACKGROUND(${SOUND_DIR}/somefile) exten => 100,n,WAITEXTEN(10) exten => 100,n,HANGUP() exten => 1,1,PLAYBACK(goodbye) .... and so on ... I've looked at the ivr.conf stuff but it's all static and all of this has to be manageable via a web interface .... meaning dumping into a DB and returning an XML response seems reasonable ... but trying to stick or modify static text files from the web interface is too much text parsing and bad things will happen ... Any thoughts or pointing me in the right direction would be appreciated. Shelby
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