search for: pbx_builtin_background

...") in new stack [Dec 12 22:53:31] WARNING[6300]: file.c:650 ast_openstream_full: File good does not exist in any format [Dec 12 22:53:31] WARNING[6300]: file.c:933 ast_streamfile: Unable to open good (format 0x4 (ulaw)): No such file or directory [Dec 12 22:53:31] WARNING[6300]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/100-096ce078 for good -- Executing [80 at outbound:18] BackGround("SIP/100-096ce078", " evening ") in new stack [Dec 12 22:53:31] WARNING[6300]: file.c:650 ast_openstream_full: File evening does not exist in any format [Dec 12 22:53:31] WARNI...

...extension for test: [Nov 25 20:44:41] WARNING[4334]: file.c:650 ast_openstream_full: File good does not exist in any format [Nov 25 20:44:41] WARNING[4334]: file.c:933 ast_streamfile: Unable to open good (format 0x8 (alaw)): No such file or directory [Nov 25 20:44:41] WARNING[4334]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-09b52260 for good -- Executing [80 at default:12] BackGround("SIP/102-09b52260", " evening ") in new stack [Nov 25 20:44:41] WARNING[4334]: file.c:650 ast_openstream_full: File evening does not exist in any format [Nov 25 20:44:41] WARNIN...

...ARNING[7634][C-00000006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-00000006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-00000006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP000005-0000000c for Fermeture -- Executing [334xx at Accueil_Phibee_HNO:4] Hangup("SIP/SIP000005-0000000c", "") in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP000005-0000000c' I understand that...

.../var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open /var/lib/asterisk/sounds/vprompts/*zip-code.wav* (format 0x8 (*alaw*)): No such file or directory [Aug 28 11:16:29] WARNING[2705]: pbx.c:5752 pbx_builtin_background: ast_streamfile failed on SIP/test1-0000000f for /var/lib/asterisk/sounds/vprompts/*zip-code.wav* Am I missing a module to translate from wav to alaw/gsm/g726/... ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/p...

...t 12 14:54:22] WARNING[13545]: file.c:385 fn_wrapper: Unable to open format wav [Oct 12 14:54:22] WARNING[13545]: file.c:991 ast_streamfile: Unable to open /var/lib/asterisk/sounds/digits/84108795/en/cents (format 0x4 (ulaw)): No such file or directory [Oct 12 14:54:22] WARNING[13545]: pbx.c:5752 pbx_builtin_background: ast_streamfile failed on SIP/134-00000087 for /var/lib/asterisk/sounds/digits/84108795/en/cents [Oct 12 14:54:22] ERROR[13545]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe -- AGI Script Executing Application: (BACKGROUND) Options: (/var/lib/asterisk/sounds/digits/841087...

...if ((confflags & CONFFLAG_ADMIN)) { /* Do admin stuff here */ } else { /* Do user menu here */ } ---------------------------------------------------------------------------------------------------- I guess to use the pbx_builtin_background ( that already implement a loop playing waitting a digit ) to play the menu , and allow Admin/User choose a option sending DTFM. And i would like to know the better way to implement that ... any hint about ? Thanks a lot, P.S : Future plans will be more complex as be able to join a new cal...

...le /var/lib/asterisk/sounds/en/vm-instructions.gsm does not exist in any format [Mar 10 00:07:45] WARNING[23934]: file.c:856 ast_streamfile: Unable to open /var/lib/asterisk/sounds/en/vm-instructions.gsm (format 0x40 (slin)): No such file or directory [Mar 10 00:07:45] WARNING[23934]: pbx.c:7138 pbx_builtin_background: ast_streamfile failed on vpb/0-3 for /var/lib/asterisk/sounds/en/vm-instructions.gsm And the built in: -- Executing [s at default:14] VoiceMail("vpb/0-0", "400") in new stack [Mar 10 00:07:36] WARNING[23934]: file.c:557 ast_openstream_full: File vm-intro does not exist in...

Hi Guys, My name is Marcias, and I am setting up for the first time an Asterisk PBX, I am learning as I go along. I have been able to download and install Asterisk, Libpri and I have been able to get Asterisk up and running. I have several questions: 1 .I can call the Asterisk server from my Xten phone and it picks up. I have 3 computers (one of them being the asterisk server) I can seem to call

...not say fmt Sep 3 11:59:22 WARNING[14350]: file.c:406 ast_filehelper: Unable to open fd on /opt/asterisk/var/lib/sounds/ifyou.wav Sep 3 11:59:22 WARNING[14350]: file.c:761 ast_streamfile: Unable to open ifyou (format SLINR): No such file or directory Sep 3 11:59:22 WARNING[14350]: pbx.c:4484 pbx_builtin_background: ast_streamfile failed on Local/541141091828@callout-204e,1 fro ifyou -- Executing BackGround("Local/xxxxxxxxxxxx@callout-204e,1", "silence/1") in new stack -- Playing 'silence/1' (language 'en') == Spawn extension (callout, xxxxxxxxxxxxx, 2) exited...