search for: codec_ulaw

...d => 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 => format_pcm.so load => format_wav.so load => format_wav_gsm.so load => res_agi.so load => res_clioriginate.so load => res_fax.so load => res_musiconhold.so load => res_timing_timerfd.so load => func_callerid.so load => func_cdr.so...

...a network with some asterisk in trunk with IAX2 and some SIP/ZAP phone connect to this *. In every call I need to use only alaw codec so in all conf file I have set disallow=all and allow=alaw. I try also to make some tuning of my environment removing unused codec and application. If I remove the codec_ulaw.so when I try to call I see this: [Oct 5 12:15:33] WARNING[16637]: chan_iax2.c:8021 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/CTM1-283 -- Hungup 'IAX2/CTM1-283' [Oct 5 12:15:33] WARNING[16637]: app_dial.c:1090 dial_exec_full: Unable to create channel...

...codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 codec_gsm.so GSM Coder/Decoder 0 codec_ulaw.so mu-Law Coder/Decoder 0 I'm assuming use=0 because the server is idle. I've got allow = all in my sip.conf file. Anyway, does anyone have an idea on how to resolv...

Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have with a telco to asterisk, don't work anymore. I don't know why!... This is the SDP portion that comes in the INVITE messages of calls

...configuration has been updated to Debian Buster and thus to Asterisk 16. Since this upgrade I have a dependency problem related to res_rtp_asterisk.so. So the old config was: [modules] autoload=no load => res_rtp_asterisk.so load => res_http_websocket.so load => chan_local.so load => codec_ulaw.so load => codec_alaw.so load => pbx_config.so load => chan_sip.so load => app_dial.so load => func_callerid.so load => func_cut.so load => func_logic.so [global] Since Asterisk 16 (Debian Buster version) I have a dependency problem, where res_rtp_asterisk.so is dependent on...

...s) seem fine. Unfortunately, when a caller listens to a Playback recording, there seems to be moments of stutter - perhaps 1 second of stutter for every 10 seconds of Playback. The stutter is not consistent at the same point of the playback file. To eliminate encoding as an issue, I have only codec_ulaw/format_pcm loaded and the recording is ulaw. I've niced down the asterisk process to -19 even though I don't see asterisk taking more than 3% cpu. Is this behavior indicative of a timing problem? loading res_timing_pthread.so makes things horribly worse. i don't believe any oth...

...sing '/etc/asterisk/extensions_custom.conf': == Found [Feb 2 08:14:46] NOTICE[32490]: pbx_ael.c:149 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. -- Reloading module 'func_odbc' (ODBC lookups) -- Reloading module 'codec_ulaw' (mu-Law Coder/Decoder) -- Reloading module 'codec_lpc10' (LPC10 2.4kbps Coder/Decoder) -- Reloading module 'codec_gsm' (GSM Coder/Decoder) -- Reloading module 'codec_g726' (ITU G.726-32kbps G726 Transcoder) -- Reloading module 'codec_g722' (ITU G...

...------------------------------------ But I've not made it very far building up just what I need: ------------------------------------ [modules] autoload=yes load => res_crypto.so load => res_features.so load => chan_iax2.so load => chan_sip.so load => codec_gsm.so load => codec_ulaw.so ------------------------------------ I get hard to track down symbol errors like: [chan_iax2.so]Aug 21 10:37:02 WARNING[16384]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_check_signature Aug 21 10:37:02 WARNING[16384]: loader.c:374 load_mo...

...[*] 6. codec_ilbc [*] 7. codec_lpc10 XXX 8. codec_speex [*] 9. codec_ulaw XXX 10. codec_zap Generic Zaptel Transcoder Codec Translator Depends on: zaptel_transcode(E), zaptel(E) I see this in the changelog: 2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming <kpfleming@digium.com>...

Greetings list, Has anyone compiled (or could point me at) a list of the minimum required modules and conf files for a very basic 1.8 deployment? We have lots of 1.4 boxes in production, and I'm currently setting up a pair of 1.8 boxes to bounce calls coming in via IAX over IPv6 over to the existing 1.4 boxes. All the new installs need to do is receive calls via IAX and send them out via

...ranslator 'lintolpc10' from format SLINR to LPC10, cost 9 [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder) == Registered translator 'adpcmtolin' from format ADPCM to SLINR, cost 1 == Registered translator 'lintoadpcm' from format SLINR to ADPCM, cost 1 [codec_ulaw.so] => (Mu-law Coder/Decoder) == Registered translator 'ulawtolin' from format ULAW to SLINR, cost 1 == Registered translator 'lintoulaw' from format SLINR to ULAW, cost 1 [codec_alaw.so] => (A-law Coder/Decoder) == Registered translator 'alawtolin' from format...

...ranslator 'lintolpc10' from format SLINR to LPC10, cost 9 [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder) == Registered translator 'adpcmtolin' from format ADPCM to SLINR, cost 1 == Registered translator 'lintoadpcm' from format SLINR to ADPCM, cost 1 [codec_ulaw.so] => (Mu-law Coder/Decoder) == Registered translator 'ulawtolin' from format ULAW to SLINR, cost 1 == Registered translator 'lintoulaw' from format SLINR to ULAW, cost 1 [codec_alaw.so] => (A-law Coder/Decoder) == Registered translator 'alawtolin' from format...

...translator 'lintolpc10' from format slin to lpc10, cost 19 [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder) == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1 == Registered translator 'lintoadpcm' from format slin to adpcm, cost 2 [codec_ulaw.so] => (Mu-law Coder/Decoder) == Registered translator 'ulawtolin' from format ulaw to slin, cost 1 == Registered translator 'lintoulaw' from format slin to ulaw, cost 1 [codec_alaw.so] => (A-law Coder/Decoder) == Registered translator 'alawtolin' from format al...

...is is not a one-way audio problem. Sometimes I can get 'her' to speak to me, other times I can't for a long time. I'm just using a very simple system to dump people into MeetMe(). Nothing fancy. I do have the following in my modules.conf: preload => format_mp3.so preload => codec_ulaw.so preload => format_pcm.so My extensions.conf looks like: [general] autofallthrough=yes static=no writeprotect=no extenpatternmatchnew=yes clearglobalvars=no [conference-calls] exten => s,1,Answer() exten => s,n,Background(welcome) exten => s,n,Background(and) exten => s,n,Backg...

...is is not a one-way audio problem. Sometimes I can get 'her' to speak to me, other times I can't for a long time. I'm just using a very simple system to dump people into MeetMe(). Nothing fancy. I do have the following in my modules.conf: preload => format_mp3.so preload => codec_ulaw.so preload => format_pcm.so My extensions.conf looks like: [general] autofallthrough=yes static=no writeprotect=no extenpatternmatchnew=yes clearglobalvars=no [conference-calls] exten => s,1,Answer() exten => s,n,Background(welcome) exten => s,n,Background(and) exten => s,n,Backg...

...em_i4l.so noload => chan_oss.so load => chan_phone.so noload => chan_sip.so noload => chan_skinny.so load => codec_a_mu.so load => codec_adpcm.so load => codec_alaw.so load => codec_g726.so load => codec_gsm.so load => codec_ilbc.so load => codec_lpc10.so load => codec_ulaw.so load => format_g726.so load => format_g729.so load => format_gsm.so load => format_h263.so load => format_ilbc.so load => format_jpeg.so load => format_pcm.so load => format_pcm_alaw.so load => format_vox.so load => format_wav.so load => format_wav_gsm.so load =&...

...cation 0 app_sayunixtime.so Say time 0 pbx_spool.so Outgoing Spool Support 1 app_macro.so Extension Macros 0 app_random.so Random goto 19 codec_ulaw.so Mu-law Coder/Decoder 0 codec_alaw.so A-law Coder/Decoder 0 app_transfer.so Transfer 0 app_verbose.so Send verbose output 0 app_setcdruserfield.so CDR user...

...I hope this gets picked up by some bug marshall ... I have downloaded (yesterday) the 1.2 branch from svn ... When running: asterisk -vvvvc loaded modules: [modules] autoload=no load => pbx_functions.so load => pbx_config.so load => codec_a_mu.so load => format_pcm_alaw.so load => codec_ulaw.so load => codec_alaw.so load => format_pcm.so load => func_uri.so ;required by app_dial and chan_sip load => res_features.so load => app_dial.so ;playback and echo apps ... load => app_playback.so load => app_echo.so load => codec_gsm.so load => format_gsm.so load =&g...

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

.../tt-monkeys.ulaw root at TX-LU-DEV:~# ls -la /var/lib/asterisk/sounds/en/jedi-extension-trick.ulaw -rw-r--r-- 1 root root 30400 2010-08-19 19:23 /var/lib/asterisk/sounds/en/jedi-extension-trick.ulaw Here is my modules.conf file (I use autoload=no to prune out certain stuff). Also, note that I have codec_ulaw.so loaded, which is, I think, the only thing required in order to play ulaw files. modules.conf: http://pastie.org/1113611 That's all the debugging information I have, if you need anything else please let me know. I get the feeling that this is related to me not loading a required module some...