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chan_sip SRTP

Добавлено: 07 июн 2019, 22:45
bublikoff
Пытаюсь подключить SIPML5
Получаю такую штуку

chan_sip.c:5989 dialog_initialize_dtls_srtp: No SRTP module loaded, can't setup SRTP session

Отдельного модуля не нашел у 16.4 ... на сколько я понял res_srtp копилиться как встроеный модуль

Содержимое modules.conf
[modules]
autoload=no

; Resources —
load => res_musiconhold.so ; Music On Hold Resource
load => res_security_log.so ; Security Event Logging
load => res_parking.so ; Call Parking Resource
load => res_rtp_asterisk.so ; Asterisk RTP Stack
load => res_rtp_multicast ; Multicast RTP Engine
load => res_pjproject.so ; PJPROJECT Log and Utility Support
load => res_sorcery_config.so ; Sorcery Configuration File Object Wizard
load => res_http_websocket.so ; HTTP WebSocket Support

; Bridges -
load => bridge_native_rtp.so ; Native RTP bridging module
load => bridge_simple.so ; Simple two channel bridging module

; PBX —
load => pbx_config.so ; Text Extension Configuration Requires N/A
load => pbx_spool.so ; Outgoing Spool Support

; Functions —
load => func_global.so ; Variable dialplan functions
load => func_callerid.so ; Gets or sets Caller*ID data on the channel.
load => func_channel.so ; Channel information dialplan functions
load => func_logic.so ; Logical dialplan functions
load => func_base64.so ; Base64 encode/decode dialplan functions
load => func_strings.so ; String handling dialplan functions
load => func_volume.so ; Technology independent volume control

; Channels —
load => chan_sip.so ; Session Initiation Protocol (SIP) - Requires res_features.so
load => chan_rtp.so ; RTP Media Channel

; Codecs —
load => codec_alaw.so ; A-law Coder/Decoder
load => codec_g722.so ; ITU G.722-64kbps G722 Transcoder
load => codec_g726.so ; ITU G.726-32kbps G726 Transcoder
load => codec_gsm.so ; GSM/PCM16 (signed linear) Codec Translat
load => codec_ulaw.so ; Mu-law Coder/Decoder

; Formats
load => format_g726.so ; Raw G.726 (16/24/32/40kbps) data
load => format_gsm.so ; Raw GSM data
load => format_pcm.so ; Raw uLaw 8khz Audio support (PCM)
load => format_wav.so ; Microsoft WAV format (8000hz Signed Linear)
load => format_wav_gsm.so ; Microsoft WAV format (Proprietary GSM)
load => format_h264.so ; Raw H.264 data (Video)

; Applications —
load => app_authenticate.so ; Authentication Application
load => app_dial.so ; Dialing Application - Requires res_features.so, res_musiconhold.so
load => app_echo.so ; Echo audio read from channel back to the channel
load => app_playback.so ; Sound File Playback Application
load => app_playtones.so ; Playtones Application
load => app_senddtmf.so ; Send DTMF digits Application
load => app_system.so ; Generic System() application
load => app_transfer.so ; Transfer caller to another extension
load => app_verbose.so ; Send arbitrary text to verbose output
load => app_exec.so ; Executes dialplan applications
load => app_stack.so ; Dialplan subroutines (Gosub, Return, etc)
load => app_queue.so ; True Call Queueing

Re: chan_sip SRTP

Добавлено: 07 июн 2019, 23:54
sasa
нет
должен быть so модуль
если нет пересобирать с нуля астериск

Re: chan_sip SRTP

Добавлено: 08 июн 2019, 03:17
bublikoff
Вы оказались правы ... астериска сбоиралась без libsrtp от чего res_srtp.so не собиралась ... пересбор дал результат