search for: app_record

Thanks for your responses - it looks like I have the following options, in order of ease: 1: Modify and recompile app_record.c Change line 471 https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L471 from status_response = "DTMF"; to status_response = dtmf_integer; Pro: Free, easy Con: Have to remember to edit module each time a new Asterisk update comes out 2: Use the Jean Aunis "m...

...uot;DTMF" (A terminating DTMF was received). > > But I need to know **what** number that DTMF was, and I can't see a > way of grabbing it after the fact. > > I can see in the code where the right variables are.. > > https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L140 > dtmf_response > > https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L166 > * \param dtmf_integer the integer value of the DTMF key received > > So,3 questions I guess: > > 1: Am I going about this the right way? (unimrcp is not an option here) >...

...gt; >> But I need to know **what** number that DTMF was, and I can't see a > >> way of grabbing it after the fact. > >> > >> I can see in the code where the right variables are.. > >> > >> https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L140 > >> dtmf_response > >> > >> https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L166 > >> * \param dtmf_integer the integer value of the DTMF key received > >> > >> So,3 questions I guess: > >> > >> 1:...

On 20 January 2018 at 23:30, Tim S <tim.strommen at gmail.com> wrote: > I have seen this take over 2 seconds before on a sluggish machine. Thanks - my host uses SSD and everything seems pretty quick, but I'll give it a 1 second pause. > you'd need to pipe that to a Google Speech API tunnel. > That's probably not something you can hack away at with simple > Asterisk

...s:mp3") in new stack -- Playing 'beep' WARNING[360468]: File translate.c, Line 128 (ast_translator_build_path): No translator path from UNKN to ULAW WARNING[360468]: File file.c, Line 218 (ast_writestream): Unable to translate to format mp3, source format ALAW WARNING[360468]: File app_record.c, Line 166 (record_exec): Problem writing frame Segmentation fault I guess this is pretty explanatory. Regards Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031028/e771aeac/attachment.ht...

using a zap fxo and zap fxs card how can I set up caller announce? like this. 1 call comes in and a prompt asks the called to identify themselves. 2 the system would then put the caller on hold and pick up the FXS and play the message for the users prompting them to hit 1 to accept the call and have it connected or hit 2 to dump the live caller to voicemail. Can this be done with * Dave

..._rewrite: I Can't write MP3 only read them. [Apr 23 17:22:36] WARNING[4494]: file.c:378 fn_wrapper: Unable to rewrite format mp3 [Apr 23 17:22:36] WARNING[4494]: file.c:1092 ast_writefile: Unable to rewrite /var/spool/asterisk/alarme/alarme-1201-200905121212.mp3 [Apr 23 17:22:36] WARNING[4494]: app_record.c:272 record_exec: Could not create file /var/spool/asterisk/alarme/alarme-1201-200905121212 I'am doing something wrong? Thanks Veja quais s?o os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com -------------- next part -------------- An HTML attachment was sc...

...sip.so load => app_authenticate.so load => app_cdr.so load => app_dial.so load => app_dumpchan.so load => app_echo.so load => app_exec.so load => app_hangup.so load => app_macro.so load => app_originate.so load => app_playback.so load => app_playtones.so load => app_record.so load => app_userevent.so load => codec_adpcm.so load => codec_alaw.so load => codec_a_mu.sothe number of modules to minimum ? load => codec_g722.so load => codec_g726.so load => codec_gsm.so load => codec_lpc10.so load => codec_ulaw.so load => format_gsm.so load =...

...application will not finish. There is a sound file, but it is empty (0 bytes). "Record(${FILE}:gsm|10|30|skip)" is used in the dial plan. After hangup the following error messages show up: NOTICE[]: channel.c:1683 ast_set_read_format: Unable to find a path from SLINR to UNKN WARNING[]: app_record.c:287 record_exec: Unable to restore read format on 'ALSA/default' Kernel 2.6.5 and Asterisk CVS 04-97-23 are installed. Some test using arecord and aplay shows that microphone and speakers are working. Any help would be appreciated. -- Stefan Tichy <asterisk@pi4tel.de>

...ot;, "/var/local/whois-messages/whois-321:wav|6|120") in new stack -- Playing 'beep' (language 'en') Jul 30 21:44:20 WARNING[4206]: file.c:910 ast_writefile: Unable to open file /var/local/whois-messages/whois-321.wav: Permission denied Jul 30 21:44:20 WARNING[4206]: app_record.c:299 record_exec: Could not create file /var/local/whois-messages/whois-321 Any ideas? Thanks...

...from debug level 5: -- AGI Script Executing Application: (Record) Options:(/tmp/asterisk/incident-3893006535:wav,) -- Playing 'beep' (language 'en') Jun 22 13:53:06 WARNING[1209214400]: file.c:856 ast_writefile: No such format 'wav,' Jun 22 13:53:06 WARNING[1209214400]: app_record.c:221 record_exec: Could not create file /tmp/asterisk/incident-3893006535 Jun 22 13:53:08 WARNING[1209214400]: file.c:464 ast_openstream: File /tmp/asterisk/incident-3893006535 does not exist in any format Jun 22 13:53:08 WARNING[1209214400]: app_agi.c:336 handle_streamfile: Unable to open /tmp/as...

...provider, and not with ZAP and a POTS line. also note that i've tried two different IAX2 providers with the same resuls. (using zap and pots lines is not preferrable in this case) one more note is that i've disabled the 'beep' before the recording starts (commented out lines in app_record.c and re-built code), just in case that was causing it -- no change in behavior, though. any thoughts on what could be causing this, or how to further troubleshoot? Regards, Steve

Thanks Jurijs, Yes, in fact I'm already using that, and it works fine. The problem here is that I cannot find a way of recording speech AND listening for a DTMF digit being pressed as an alternative. That's where the problem lies. J.

...#39;m using cvs-head (2004-01-17) and chan_capi 0.3.5 (with my patches to make chan_capi compile with asterisk cvs-head) and a diva quad-bri isdn card. other things work well with my setup (dial in, dial out, app_meetme) and sound recordings from sip channels. the problem is with voicemail and app_record, where only a distorted sound can be heard in the recording if one shouts into the microphone of the telephone. has anybody had the same problem or can confirm this issue? regards frank

...check_header: Not a wav file 49 Oct 16 12:49:41 WARNING[8581]: file.c:436 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/beep.wav Oct 16 12:49:41 WARNING[8581]: file.c:824 ast_streamfile: Unable to open beep (format ulaw): No such file or directory Oct 16 12:49:41 WARNING[8581]: app_record.c:247 record_exec: ast_streamfile failed on IAX2/308-4 I'm getting the same issue with voicemail when that application tries to play the beep. Don't think it'a actually a problem with the playing back of WAV files as such, since ext. 666: '666' => 1. Answer()...

Hello everyone! So now I'm testing with chan_sip and I discovered, that I can make calls, even if they're only listed as active channels. But JACK just emmits white noise, with a highger frequency than 8kHz, in my believe. A call with app_record shows, that the signal is clear and very ear friendly though. So how to proceed from here? What can I do to help someone look into app_jack, as I can't beyond the simplest of levels. Kindly yours Julien -------- Music was my first love and it will be my last (John Miles) ======...

Is there a prebuild module/dialplan which gives me a nice interface to recording messages? Assuming I can't use the voicemail command, I need to offer users a way to record, playback, erase, rerecord, etc. I can probably do it through dialplan but it feels like I'm reinventing the wheel. Thanks, MD

...k accept beep.gsm), what could be? [May 5 00:44:16] WARNING[2262]: file.c:663 ast_openstream_full: File beep does not exist in any format [May 5 00:44:16] WARNING[2262]: file.c:958 ast_streamfile: Unable to open beep (format 0x4 (ulaw)): No such file or directory [May 5 00:44:16] WARNING[2262]: app_record.c:285 record_exec: ast_streamfile failed on DAHDI/1-1 Regards Bilal

...ven when ICE is not enabled (Reported by Joshua Colp) * ASTERISK-25427 - Callerid change does not always emit NewCallerid AMI event (Reported by Ivan Poddubny) * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) * ASTERISK-25410 - app_record: RECORDED_FILE variable not being populated (Reported by Kevin Harwell) * ASTERISK-25394 - pbx: Incorrect device and presence state when changing hint details (Reported by Joshua Colp) * ASTERISK-25396 - chan_sip: Extremely long callerid name causes invalid SIP (Reported by Walt...

...hen divided in multiple recordings (Reported by Nuno Borges) Improvements made in this release: ----------------------------------- * ASTERISK-24283 - [patch]Microseconds precision in the eventtime column in the cel_odbc module (Reported by Etienne Lessard) * ASTERISK-24530 - [patch] app_record stripping 1/4 second from recordings (Reported by Ben Smithurst) * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded lookups (Reported by Birger "WIMPy" Harzenetter) For a full list of changes in this release, please see the ChangeLog: http://downloads.asteri...