search for: res_speech

...n 1.4.12 against the exact same system (without restarting the host) and that built fine. My host's details: glibc-2.5.1, gcc-4.2.1, binutils-2.17. Here's the last few lines of output before it crashed: [CC] res_smdi.c -> res_smdi.o [LD] res_smdi.o -> res_smdi.so [CC] res_speech.c -> res_speech.o [LD] res_speech.o -> res_speech.so [CC] chan_agent.c -> chan_agent.o [LD] chan_agent.o -> chan_agent.so [CC] chan_iax2.c -> chan_iax2.o [CC] iax2-parser.c -> iax2-parser.o [CC] iax2-provision.c -> iax2-provision.o [LD] chan_iax2.o i...

Dear All Can you let me know is this possible to if we are using Asterisk version 1.4 or 1.6 for incoming voicemail we can send as email in text formta. Means voice mesage converted into text message & send it to resp. email ids. is this possible. If yes. we can do the same with help of Asterisk or we require expertnal application need to isntall/integrate to work for speech to test. Please

...============================================== But when I try to load the lumenvox module the entire pbx is killed, the message is --------------------------------------------------------------------------------------------------------------------------------------------------------- NOTICE[9278]: res_speech_lumenvox.c:841 load_module: Lumenvox SRE Connector module Copyright (C) 1999-2007 Digium, Inc. NOTICE[9278]: res_speech_lumenvox.c:842 load_module: This module is supplied under a commercial license granted by Digium, Inc. == Parsing '/etc/asterisk/lumenvox.conf': Found -- Using serve...

...002f8000 r-xp 00000000 08:02 723040 /usr/lib/asterisk/modules/res_fax.so 002f8000-002f9000 rw-p 00011000 08:02 723040 /usr/lib/asterisk/modules/res_fax.so 002f9000-002fb000 r-xp 00000000 08:02 723055 /usr/lib/asterisk/modules/res_speech.so 002fb000-002fc000 rw-p 00001000 08:02 723055 /usr/lib/asterisk/modules/res_speech.so 002fc000-0030b000 r-xp 00000000 08:02 723047 /usr/lib/asterisk/modules/res_odbc.so 0030b000-0030c000 rw-p 0000f000 08...

...g with slashes. - Remove files that are no longer updated - app_voicemail: Add AMI event for mailbox PIN changes. - app_queue.c: Emit unpause reason with PauseQueueMember event. - bridge_simple: Suppress unchanged topology change requests - res_pjsip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is...

...g with slashes. - Remove files that are no longer updated - app_voicemail: Add AMI event for mailbox PIN changes. - app_queue.c: Emit unpause reason with PauseQueueMember event. - bridge_simple: Suppress unchanged topology change requests - res_pjsip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is...

...g with slashes. - Remove files that are no longer updated - app_voicemail: Add AMI event for mailbox PIN changes. - app_queue.c: Emit unpause reason with PauseQueueMember event. - bridge_simple: Suppress unchanged topology change requests - res_pjsip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is...

...g with slashes. - Remove files that are no longer updated - app_voicemail: Add AMI event for mailbox PIN changes. - app_queue.c: Emit unpause reason with PauseQueueMember event. - bridge_simple: Suppress unchanged topology change requests - res_pjsip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is...

...g with slashes. - Remove files that are no longer updated - app_voicemail: Add AMI event for mailbox PIN changes. - app_queue.c: Emit unpause reason with PauseQueueMember event. - bridge_simple: Suppress unchanged topology change requests - res_pjsip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is...

...g with slashes. - Remove files that are no longer updated - app_voicemail: Add AMI event for mailbox PIN changes. - app_queue.c: Emit unpause reason with PauseQueueMember event. - bridge_simple: Suppress unchanged topology change requests - res_pjsip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is...

...jsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29040 - res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) * ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) * ASTERISK-27273 - app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/proc...

...jsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29040 - res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) * ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) * ASTERISK-29034 - Lastpause of realtime members is reseting (Reported by Evandro C��sar Arruda) * ASTERISK-...

...jsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29040 - res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) * ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) * ASTERISK-29034 - Lastpause of realtime members is reseting (Reported by Evandro C��sar Arruda) * ASTERISK-...

Hi, I compiled Asterisk 10.7.0 with gcc-4.5.3 and at runtime I'm getting these warnings: loader.c: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref loader.c: Error loading module 'app_stack.so': /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister loader.c: Error loading module

...te.so res_convert.so res_crypto.so res_fax.so res_jabber.so res_limit.so res_monitor.so res_musiconhold.so res_mutestream.so res_phoneprov.so res_realtime.so res_rtp_asterisk.so res_rtp_multicast.so res_security_log.so res_smdi.so res_speech.so res_stun_monitor.so res_timing_dahdi.so res_timing_pthread.so res_timing_timerfd.so res_calendar.so Asterisk*CLI> module load ch chan_agent.so chan_bridge.so chan_gtalk.so chan_iax2.so chan_jingle.so chan_local.so chan_mgcp.so chan_mu...

...== Manager registered action PauseMonitor == Manager registered action UnpauseMonitor res_monitor.so => (Call Monitoring Resource) == WebSocket registered sub-protocol 'echo' res_http_websocket.so => (HTTP WebSocket Support) res_crypto.so => (Cryptographic Digital Signatures) res_speech.so => (Generic Speech Recognition API) == AGI Command 'answer' registered == AGI Command 'asyncagi break' registered == AGI Command 'channel status' registered == AGI Command 'database del' registered == AGI Command 'database deltree' registered...

...adm.so.1...done. Loaded symbols for /usr/lib/libadm.so.1 Reading symbols from /usr/lib/libcryptoutil.so.1...done. Loaded symbols for /usr/lib/libcryptoutil.so.1 Reading symbols from /usr/lib/libdoor.so.1...done. Loaded symbols for /usr/lib/libdoor.so.1 Reading symbols from /opt/asterisk/lib/modules/res_speech.so...done. Loaded symbols for /opt/asterisk/lib/modules/res_speech.so Reading symbols from /opt/asterisk/lib/modules/res_agi.so...done. Loaded symbols for /opt/asterisk/lib/modules/res_agi.so Reading symbols from /opt/asterisk/lib/modules/res_clioriginate.so...done. Loaded symbols for /opt/asterisk...

...ymbols for /usr/lib/libadm.so.1 > Reading symbols from /usr/lib/libcryptoutil.so.1...done. > Loaded symbols for /usr/lib/libcryptoutil.so.1 > Reading symbols from /usr/lib/libdoor.so.1...done. > Loaded symbols for /usr/lib/libdoor.so.1 > Reading symbols from /opt/asterisk/lib/modules/res_speech.so...done. > Loaded symbols for /opt/asterisk/lib/modules/res_speech.so > Reading symbols from /opt/asterisk/lib/modules/res_agi.so...done. > Loaded symbols for /opt/asterisk/lib/modules/res_agi.so > Reading symbols from /opt/asterisk/lib/modules/res_clioriginate.so...done. > Loaded...

...jsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29040 - res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) * ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) * ASTERISK-29034 - Lastpause of realtime members is reseting (Reported by Evandro C��sar Arruda) * ASTERISK-...

...sterisk/agi-bin ; done make[1]: Leaving directory `/usr/src/asterisk-1.4.20.1/agi' make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/res' for x in res_adsi.so res_agi.so res_clioriginate.so res_convert.so res_features.so res_indications.so res_monitor.so res_musiconhold.so res_smdi.so res_speech.so; do /usr/bin/install -c -m 755 $x /usr/lib/asterisk/modules ; done make[1]: Leaving directory `/usr/src/asterisk-1.4.20.1/res' make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/channels' for x in chan_agent.so chan_iax2.so chan_local.so chan_mgcp.so chan_oss.so chan_phone.so chan_s...