MC, thank you for adding this macro. it seems to work fine. for my current purposes i just created silent wavs until i need the complete voice instructions. ________________________________ From: "freeswitch-users-request@lists.freeswitch.org" <freeswitch-users-request@lists.freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Saturday, January 21, 2012 9:32 AM Subject: FreeSWITCH-users Digest, Vol 67, Issue 195 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to     freeswitch-users@lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to     freeswitch-users-request@lists.freeswitch.org You can reach the person managing the list at     freeswitch-users-owner@lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics:   1. Re: mod_rtmp + Akamai stream push (Anthony Minessale)   2. Re: announce count to user only entering conference - Parse       Error (Michael Collins)   3. Re: G.722 negotiation issue with AVM Fritzbox (Brian West) I don't recall any initiative being done to facilitate what you describe but its open source code so nothing's impossible.   On Thu, Jan 19, 2012 at 4:08 PM, Jock McKechnie <jock.mckechnie@gmail.com> wrote: Greetings wise(r) FreeSWITCH users;

We've recently started using FreeSWITCH and mod_rtmp to do RTMP -> SIP media conversion (audio only) and have been utterly delighted by how easy and (so far) reliable this has been to both set up and run on a daily basis. One of my superiors has gotten all excited and wonders if we can use the same mechanism to do a stream push to Akamai for content delivery amongst many thousands of RTMP clients (think radio broadcast).

It appears that the FreeSWITCH mod_rtmp module is _not_ capable of this - it can send a fresh RTMP stream to a registered client, but it cannot arbitrarily start pushing RTMP to an Akamai EntryPoint. Am I correct in this assertion, or am I misunderstanding the documentation (and google hits) available?

My thanks to all;

 - Jock

_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com

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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

-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 I would use a phrase macro for this. In fact, this seems like such an obvious thing to do that I added a phrase macro to conf/lang/en/ivr/sounds.xml. Update to the latest latest like two seconds ago git and you will have it available. (You'll need to copy it over from the source directory to your install directory.) The new macro is called other_callers_in_conf. In your example the extension would be this: <extension name="room1">       <condition field="destination_number" expression="^301$">         <action application="set" data="conference_user_list=|"/>        <action application="set" data="conf_count=${conference 301-127.0.0.1 list count}"/>         <action application="log" data="INFO ${conf_count}"/>   <!-- play current conference count-->        <action application="playback" data="phrase:other_callers_in_conf:${conf_count}"/>         <action application="conference" data="301-127.0.0.1@default"/>        <action application="transfer" data="300 xml default" />    </condition> </extension> The phrase macro does all the work. NOTE: I need to record a few new phrase files and then tweak the macro. In the meantime I encourage you to look at the macro that I added and tailor it to your needs. -MC On Thu, Jan 19, 2012 at 12:45 PM, Rodney <notlikeme75@yahoo.com> wrote: Okay, I finally figured out how to do this hopefully since I know what conference they are entering and all seems to work accept when the room is empty. I am putting phrases before and after the say to make it more understandable to "some people" what the number means. 

scenario:

When I enter a room with people the system says "there are"   say count  "other callers" because it is a count without them in the room yet, unlike the normal caller control to call the announce extension when they are already in the room.

this works fine when there is at least 1 other caller in the room. 

but when there isn't another caller and the room is empty. it says "there are" say count is a parse error  so no digit read  (need it to say zero) then it say "other callers" then says my normal you are in the conference alone message which for now i have changed to say "zero" so it says sort of backwards "there are" "other callers' "zero" 

ERR] mod_say.c:130 Parse Error!

 Could someone please help me understand why this Parse error could be happening on zero count. I would like to fix it. thank you.

<extension name="room1">       <condition field="destination_number" expression="^301$"> <action application="set" data="conference_user_list=|"/>   <!-- play current conference count-->         <action application="set" data="conf_count=${conference 301-127.0.0.1 list count}"/>         <action application="log" data="INFO ${conf_count}"/>         <action application="say" data="en number pronounced ${conf_count}"/>         <action application="playback" data="C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/othercallers.wav"/>    <!-- add this call to the conference --> <action application="conference" data="301-127.0.0.1@default"/>        <action application="transfer" data="300 xml default" />     </condition></extension>    

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What is your codec selection set to in FreeSWITCH and what rev of FreeSWITCH are you running? On Jan 20, 2012, at 4:09 PM, Roland Hänel wrote: OK you're right, I read the RfC and in fact a zero port means that the

stream is rejected. However the whole "negotiation" is still broken. If FS rejects the incorrect G.722, OK, but what about all the other (perfectly valid) codecs in the request (PCMA, PCMU)?

Because FS only rejects G.722, the call is doomed to fail, even though it could work nicely if FS just selected one of the acceptable offers.

Roland

--  Brian West  FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266  Fax:   +1 (918) 420-9267 brian@freeswitch.org 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