September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

PCMU ringback gets weird if you specify the file extension?
by Jock McKechnie 03 Sep '10

03 Sep '10

Hey; So with early media happy, I'm working on various media files that my company wants played back. Can anyone shed any light on why if I do this: <action application='set' data='ringback=sounds/early-media.PCMU'/> The audio is garbage. But if I do this: <action application='set' data='ringback=sounds/early-media'/> The audio is perfect, even though the console clearly shows it is playing _the same file_: mod_native_file.c:94 Opening File [/usr/local/freeswitch/sounds/early-media.PCMU] 8000hz I've got a workaround, so I'm not bothered, but... it seems awfully strange that specifying the extension confuses it when they are playing the same file. I'm currently running git.head as of Wednesday. Thanks; - Jock

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Get Errors, Warning, Notices in events?
by Robert Hadley 03 Sep '10

03 Sep '10

Hi FS users, Is there a way to use mod_event_socket to listen for the different console log level messages (alarms, errors, warning, or notices) as events? Thanks, Robert

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What is the purpose of SWITCH_SIG_BREAK?
by jesse 03 Sep '10

03 Sep '10

Could FS developer please illustrate it to me? in what kind of case FS will send out SWITCH_SIG_BREAK instead of SWITCH_SIG_KILL. Another question is what is the difference between channel_on_execute and channel_on_softexecute? thanks! -jesse

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Handling US and UK English with say application
by Pete Kelly 03 Sep '10

03 Sep '10

Hi I currently have a lua script which is reading out an account balance like this: session:execute("say", "en number pronounced 12345"); I have 2 sets of sound files, one for US english and another for UK english, however the say application only appears to allow one directory to be configured for "en" and I can't see any way of modifying this on the fly so I can switch between sound files for each user. Does anybody know of a solution to this? Pete

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Book/eBook about FreeSwitch.
by Rodrigo Lang 03 Sep '10

03 Sep '10

Hi everyone. I begin study about FreeSwitch, I work with Asterisk to five years, so my experience with sip is good. But it's hard in Brazil to find a good documentation. Has the wiki.freeswitch and I know, yet I looking for something more deep about the FS. Have some book or ebook to buy? Or something free. Anything about is good and helpful. Thanks a lot and sorry for the bad english! :) Regards, -- Rodrigo Lang http://rodrigorecipes.blogspot.com/<http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html>

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Re: [Freeswitch-users] mod_freetdm dialing delay on analog line
by Joseph L. Casale 03 Sep '10

03 Sep '10

>Wondered why there would be an approximate 6-7 second delay in calling analog/POTS >via Sangoma USB100. Have used a TDM400P and there's a delay, but up to 7 seconds seems >excessive. Mark, You ever figure this one out?

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Channels and calls lost in SQLite
by João Mesquita 03 Sep '10

03 Sep '10

Hello guys, I have a problem that started to happen after I upgraded to git head last night. No configuration changes were made and this is the intriguing part. I have SQLite being mounted on tmpfs on busy Dell 1950s. After some time on the busy moments of the day, FS starts to not delete channels/calls from SQLite and leaves very little trace for me to track. I grep'ed the logs for SQL errors and found nothing.. The only part of the log that I could spot and that called my attention was this line that repeated several times during the periods where the problem occurs: switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "queued event at a lower priority %d/%d!\n", index, was); 11:23:13.977874 [WARNING] switch_event.c:1353 queued event at a lower priority 1/0! Which is printed from the following code: for (index = (*event)->priority; index < 3; index++) { int was = (*event)->priority; if (switch_queue_trypush(EVENT_QUEUE[index], *event) == SWITCH_STATUS_SUCCESS) { if (index != was) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "queued event at a lower priority %d/%d!\n", index, was); } goto end; } >From what I could understand on the code, seems like there are 3 event priority queues and that message occurs when queue 0 is full. How exactly that relates to the problem, I am not sure so I see myself here lost in translation. Could the core_event_handler not receive some messages therefore not execute the sql statements? If that's the case, maybe I lost some log line telling me that we were loosing events? I hope someone can shed some light. Thank you, João Mesquita

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Re: [Freeswitch-users] RecordFile returns prematurely
by Dan 03 Sep '10

03 Sep '10

Brian, thanks for your response. I'm confused why my script is exiting at all as recordFile should block until the call is hung up by the far end, it does this 99% of the time. As a test I have put a log message right after the recordFile line, this gets printed out, so the script is continuing on after the recordFile call is returning (which is should not as the far end has not disconnected) As I understand autoHangup is used to keep processing the dial plan when the script is done. In my case the script should not have finished at all, the call is still up, nobody has sent any hangup yet. What would make recordFile return internally with no arguments except the file name? Is there some buffer, lock, rtp timeout that could be causing this? From: "Brian West" <brian(a)freeswitch.org> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Wednesday, September 1, 2010 6:59:20 PM Subject: Re: [Freeswitch-users] RecordFile returns prematurely because your script exited and you didn't set the autoHangup to false. See mod_lua wiki page. /b On Sep 1, 2010, at 6:53 PM, Dan wrote: > Right, session.recordFile returns, its the last line in my lua|javascript script, and the dial plan has nothing left to execute so it hangs up. The question is why is session.recordFile returning when it should be blocking until the call is hung up (remotely). There is no bye or media disconnect from the far end, the record call is just returning and freeswitch is then hanging the call up and sending a bye to the far end client as there's nothing left to do. > ----- Original Message ----- > From: "Brian West" <brian(a)freeswitch.org> > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> > Sent: Wednesday, September 1, 2010 4:32:48 PM > Subject: Re: [Freeswitch-users] RecordFile returns prematurely > > Read this line. > > /b _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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how to register a gateway to freeswitch
by Octavio Duarte 03 Sep '10

03 Sep '10

Hello everybody I have FreeSwitch running and i have done some call between extensions, i bought the book on amazon its a great book but its focus to make a pbx , i could not find an answer to my problem, i have a Quintum and 2 land lines connected to it. My quintum does not have a fixed IP, i wanna use my Quintum as a gateway to sent call to PSTN, but i cant beacause i dont know the ip address. i wanna know if is there a way to register the gateway to freeswitch? or maybe i can register the quintum as a endpoin but how can i sent the dialed number()destination_number) to that endpoint or user? so i either want to use mod distributor (simple round-robin) and Limit on it, beacuse all examples i have seen is for using with gateways! I have read mod sofia and almost the whole book but i can find any example for doing this! Thanks in advance for everyone in this project that is amazing!

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mod_freetdm dialing delay on analog line
by Mark Maly 02 Sep '10

02 Sep '10

Hello, Wondered why there would be an approximate 6-7 second delay in calling analog/POTS via Sangoma USB100. Have used a TDM400P and there's a delay, but up to 7 seconds seems excessive. Any comments? 2010-09-02 16:56:28.586905 [DEBUG] mod_freetdm.c:427 FreeTDM/1:1/xxxxxxx CHANNEL ROUTING 2010-09-02 16:56:28.586905 [DEBUG] switch_ivr_originate.c:66 (FreeTDM/1:1/xxxxxxx) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-09-02 16:56:28.586905 [DEBUG] switch_core_session.c:1039 Send signal FreeTDM/1:1/xxxxxxx [BREAK] 2010-09-02 16:56:28.586905 [DEBUG] switch_core_state_machine.c:341 (FreeTDM/1:1/xxxxxxx) State ROUTING going to sleep 2010-09-02 16:56:28.586905 [DEBUG] switch_core_state_machine.c:314 (FreeTDM/1:1/xxxxxxx) Running State Change CS_CONSUME_MEDIA 2010-09-02 16:56:28.586905 [DEBUG] switch_core_state_machine.c:360 (FreeTDM/1:1/xxxxxxx) State CONSUME_MEDIA 2010-09-02 16:56:28.586905 [DEBUG] switch_core_state_machine.c:360 (FreeTDM/1:1/xxxxxxx) State CONSUME_MEDIA going to sleep 2010-09-02 16:56:30.264549 [DEBUG] ftmod_analog.c:720 [s1c1][1:1] Detected tone DIAL on 1:1 2010-09-02 16:56:30.284864 [DEBUG] ftdm_io.c:3041 [s1c1][1:1] Generating DTMF [xxxxxxx] 2010-09-02 16:56:30.624887 [DEBUG] ftdm_io.c:2865 [s1c1][1:1] Queuing DTMF 2 2010-09-02 16:56:31.224598 [DEBUG] ftdm_io.c:2865 [s1c1][1:1] Queuing DTMF 3 2010-09-02 16:56:34.364947 [DEBUG] ftmod_analog.c:413 [s1c1][1:1] Changed state from DIALING to UP Thanks, Mark

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