Taking over the session while it's parked like that from your code and asking it to play a file is making it do 2 things at once. The session's thread is already busy in the park loop so you would have to change it's state to something passive like soft_execute so it's thread was not doing anything or send it a message so the park loop would pick it up by using the uuid_broadcast app::arg
there are variables you can set that begin group_confirm_* that you can set to do this all without even using a script. http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables On Fri, Aug 28, 2009 at 3:53 PM, Phillip Jones <[email protected]> wrote: > The dialplan is is kicking off the C# script: > > <condition field="destination_number" expression="^(12125553666)$"> > * <action application="managed" data="MyFSProjects.ProcessCall"/>* > </condition> > > I launch a thread and specify the uuid to originate: > > fsApi.Execute("originate", > string.Format("{{ignore_early_media=true,absolute_codec_string='PCMU'}}[* > origination_uuid={0}*,origination_caller_id_number={1}]sofia/gateway/broadvox/{2} > &park", *call.Uuid*, call.OutgoingCallerID, call.NumberToDial)); > > in the callback i try and play the wav file: > > FreeSWITCH.Native.ManagedSession blegSession = new > FreeSWITCH.Native.ManagedSession(callID); > > if (blegSession.Ready() && blegSession.mediaReady()) > { > blegSession.Answer(); > blegSession.Execute("playback", > "prompts/press-1-to-accept-call-from.wav"); > } > > I go on to bridge the leg a and b - that bits works fine. > > > > On Fri, Aug 28, 2009 at 4:40 PM, Anthony Minessale< > [email protected]> wrote: > > nevermind, > > > > I didn't look closely at your code. > > > > how did you end up with blegSession ? > > Are you running this script from the CLI or via the application > interface. > > > > > > > > > > > > > > > > On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale > > <[email protected]> wrote: > >> > >> the L16 codec is used because the wav files is raw PCM audio > >> which is being encoded to PCMU > >> > >> Have you tried doing this to a locally registered phone? > >> > >> > >> > >> On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones <[email protected]> > >> wrote: > >>> > >>> thanks for the reply. > >>> > >>> Tried this and exactly the same result. The prompt is slowed - as if > >>> the person is drunk. > >>> > >>> I do notice that the originating call SDP shows: > >>> > >>> a=rtpmap:0 PCMU/8000 > >>> > >>> but the play tries to use: > >>> > >>> EXECUTE sofia/external/6093693828 > >>> playback(prompts/press-1-to-accept-call-from.wav) > >>> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec > >>> Activated l...@8000hz 1 channels 20ms > >>> > >>> > >>> Should the codec be p...@8000hz as opposed to l...@8000hz? If so of > >>> there a way of forcing this codec for play. > >>> > >>> Thanks > >>> > >>> > >>> Phillip > >>> > >>> > >>> Full code & logs > >>> http://pastebin.freeswitch.org/10147 > >>> http://pastebin.freeswitch.org/10150 > >>> > >>> > >>> > >>> On Thu, Aug 27, 2009 at 4:30 PM, Anthony > >>> Minessale<[email protected]> wrote: > >>> > try wedging {ignore_early_media=true} before the first [ in your dial > >>> > string > >>> > and eliminate the code waiting for answer. > >>> > > >>> > > >>> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <[email protected] > > > >>> > wrote: > >>> >> > >>> >> Hi there, > >>> >> > >>> >> I know there are other ways of doing this. I am just trying get to > >>> >> know have fun with the FreeSWITCH API. > >>> >> > >>> >> I am using originate and park a call: > >>> >> > >>> >> fsApi.Execute("originate", > >>> >> > >>> >> > >>> >> > string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2} > >>> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial)); > >>> >> > >>> >> That works great, the phone rings. I want to play something to this > >>> >> called party when they pick up. > >>> >> > >>> >> while (!blegSession.answered()) > >>> >> { > >>> >> Log.WriteLine(LogLevel.Alert, > >>> >> "Inside::CallReturns:!Session.answered::Loop"); > >>> >> blegSession.sleep(500, 1); > >>> >> } > >>> >> > >>> >> string promptFile = "prompts/whisper.wav"; > >>> >> blegSession.StreamFile(promptFile, 0); > >>> >> > >>> >> > >>> >> This works - but the audio is choppy and slow. > >>> >> > >>> >> Is there something I need to do to that parked call before streaming > >>> >> that > >>> >> file? > >>> >> > >>> >> > >>> >> Thanks for any input. > >>> >> > >>> >> Phil > >>> >> > >>> >> _______________________________________________ > >>> >> FreeSWITCH-users mailing list > >>> >> [email protected] > >>> >> 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:[email protected]<msn%[email protected]> > >>> > GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> > >>> > IRC: irc.freenode.net #freeswitch > >>> > > >>> > FreeSWITCH Developer Conference > >>> > sip:[email protected]<sip%[email protected]> > >>> > iax:[email protected]/888 > >>> > googletalk:[email protected]<googletalk%3aconf%[email protected]> > >>> > pstn:213-799-1400 > >>> > > >>> > _______________________________________________ > >>> > FreeSWITCH-users mailing list > >>> > [email protected] > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > > >>> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> > http://www.freeswitch.org > >>> > > >>> > > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> [email protected] > >>> 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:[email protected] <msn%[email protected]> > >> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:[email protected] <sip%[email protected]> > >> iax:[email protected]/888 > >> googletalk:[email protected]<googletalk%3aconf%[email protected]> > >> pstn:213-799-1400 > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:[email protected] <msn%[email protected]> > > GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:[email protected] <sip%[email protected]> > > iax:[email protected]/888 > > googletalk:[email protected]<googletalk%3aconf%[email protected]> > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > [email protected] > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > 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:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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