1, OK,
2. Right now I have a php script calling bgapi via and event socket with the call parameters. Is that what you mean? If not, can you give me a pointer? I had assumed that every time I called bgapi it with the script in it, it would get it's own copy. Regards, ________________________________ From: [email protected] [mailto:[email protected]] On Behalf Of Anthony Minessale Sent: 17 February 2009 20:38 To: [email protected] Subject: Re: [Freeswitch-users] Big delays in playing audio files 1) turn off crash protection. 2) you cant manipulate more that one call per script, design the script to be run from the application interface so you originate the call with the api interface and transfer the call to the script so each one has it's own copy of the script. On Tue, Feb 17, 2009 at 2:30 PM, Nik Middleton <[email protected]> wrote: I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my office number (using latest trunk) 2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259 handle_fatality() Caught signal 11 for unmapped thread!Aborted (core dumped) Also then I had tone detect on, I'd often get this freeswitch: src/switch_ivr_async.c:1328: switch_ivr_tone_detect_session: Assertion `read_codec != ((void *)0)' failed. Hardware, HP DL360 G4. Centos 5.2, 4 GB ram. -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Nik Middleton Sent: 17 February 2009 20:11 To: [email protected] Subject: Re: [Freeswitch-users] Big delays in playing audio files Pretty much I haven't included the on-event hooks as it never gets to the point where they're called. Only other thing is the dial it's self, attached below. However, I notice in the default dial plan, if I call extension 1001 from 1000 it takes about 2-3 seconds for the phone to ring. Is that normal? //build dial string var dial_string = "{absolute_codec_string=PCMA," + "accountcode=" + account_code + ",ignore_early_media=true" + " ,origination_caller_id_number=" + caller_id + ",originate_timeout=25}" + "sofia/gateway/" + "mygateway/" + dial_num + "' " var first_session = new Session(dial_string); // Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; } else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; } disp_call() exit(); } //set the on_hangup function to be called when this session is hungup first_session.setHangupHook(on_hangup,"hup"); -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Michael Collins Sent: 17 February 2009 19:25 To: [email protected] Subject: Re: [Freeswitch-users] Big delays in playing audio files Is this the entire script?! -MC On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton <[email protected]> wrote: > if (first_session.ready()) { > > console_log("notice","Session state=[" + > first_session.state + "] \n"); > > > > consoleLog("NOTICE", "ready: Start DTMF\n"); > > > > first_session.execute("start_dtmf"); > > first_session.answer( ); > > > > Disposition = "ANS"; > > > > first_session.sleep(1500); > > console_log("notice", "Playing message: " + > recording + "\n"); > > first_session.streamFile(recording, on_event); > > > > if (first_session.ready()) { > > consoleLog("err", "ready: Waiting for input\n"); > > first_session.streamFile("4.wav",on_event, "dtmf"); > > consoleLog("err", "ready: Timeout on input\n"); > > first_session.execute("stop_tone_detect"); > > > > //disp_call() > > first_session.hangup() > > first_session.execute("sleep", "2000"); > > consoleLog("NOTICE", "EXITING\n"); > > exit(); > > } > > } > > > > ________________________________ > > From: [email protected] > [mailto:[email protected]] On Behalf Of Anthony > Minessale > Sent: 17 February 2009 18:34 > To: [email protected] > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > > > we would need to see your script. > > On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton > <[email protected]> wrote: > > Having spent the last week developing a small js app, I ran some tests > today. With just 5 calls going on, I'm seeing huge delays from when the call > is answered to when the audio file is played. Sometimes it doesn't even > play at all!! > > > > Example 3 calls and the matching playbacks > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > > > That's 22 seconds for the first one!! > > > > Anyone any ideas as to what's going on here? > > > > Regards > > > > > > _______________________________________________ > 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/ > > AIM: anthm > MSN:[email protected] <mailto:msn%[email protected]> > GTALK/JABBER/PAYPAL:[email protected] <mailto:paypal%[email protected]> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[email protected] <mailto:sip%[email protected]> > iax:[email protected]/888 > googletalk:[email protected] <mailto: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 _______________________________________________ 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/ AIM: anthm MSN:[email protected] <mailto:msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected] <mailto:paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <mailto:sip%[email protected]> iax:[email protected]/888 googletalk:[email protected] <mailto: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
