micmac1
commented
2 years ago Hi Dost,
res_crypto gets installed together with the main asterisk package nowadays. Check your modules directory, /usr/lib/asterisk/modules, it should be there.
Seb
Closed dmsherazi closed 2 years ago
micmac1
commented
2 years ago Hi Dost,
res_crypto gets installed together with the main asterisk package nowadays. Check your modules directory, /usr/lib/asterisk/modules, it should be there.
Seb
dmsherazi
commented
2 years ago @micmac1 yes its available , Thanks for this info. Can you please help understand why the DTLS failure occurs? and how can solve it ?
dmsherazi
commented
2 years ago I checked the running modules and I got the following result
SHAULA-001*CLI> module show
Module Description Use Count Status Support Level
acl Named ACL system 2 Running core
app_blind_transfer.so Blind transfer channel to the given dest 0 Running extended
app_confbridge.so Conference Bridge Application 1 Running core
app_dial.so Dialing Application 0 Running core
app_echo.so Simple Echo Application 0 Running core
app_macro.so Extension Macros 0 Running deprecated
app_page.so Page Multiple Phones 0 Running core
app_playback.so Sound File Playback Application 0 Running core
app_playtones.so Playtones Application 0 Running core
app_stack.so Dialplan subroutines (Gosub, Return, etc 0 Running core
app_system.so Generic System() application 0 Running core
app_verbose.so Send verbose output 0 Running core
bridge_builtin_features.so Built in bridging features 1 Running core
bridge_simple.so Simple two channel bridging module 0 Running core
bridge_softmix.so Multi-party software based channel mixin 0 Running core
ccss Call Completion Supplementary Services 2 Running core
cdr CDR Engine 1 Running core
cel CEL Engine 1 Running core
chan_console.so Console Channel Driver 0 Running extended
chan_pjsip.so PJSIP Channel Driver 0 Running core
chan_rtp.so RTP Media Channel 0 Running core
codec_g729.so g729 Coder/Decoder, based on Bcg729 0 Running unknown
codec_gsm.so GSM Coder/Decoder 0 Running core
codec_opus_open_source.so Opus Coder/Decoder 0 Running unknown
codec_ulaw.so mu-Law Coder/Decoder 0 Running core
dnsmgr DNS Manager 2 Running core
dsp DSP 1 Running core
enum ENUM Support 1 Running core
extconfig Configuration 15 Running core
features Call Features 1 Running core
format_g729.so Raw G.729 data 0 Running core
format_gsm.so Raw GSM data 0 Running core
format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 Running core
func_callerid.so Party ID related dialplan functions (Cal 0 Running core
func_curl.so Load external URL 1 Running core
func_logic.so Logical dialplan functions 0 Running core
func_pjsip_aor.so Get information about a PJSIP AOR 0 Running core
func_pjsip_contact.so Get information about a PJSIP contact 0 Running core
func_pjsip_endpoint.so Get information about a PJSIP endpoint 0 Running core
func_strings.so String handling dialplan functions 0 Running core
func_timeout.so Channel timeout dialplan functions 0 Running core
http Built-in HTTP Server 5 Running core
indications Indication Tone Handling 1 Running core
logger Logger 1 Running core
manager Asterisk Manager Interface 1 Running core
pbx_config.so Text Extension Configuration 0 Running core
plc PLC 1 Running core
res_agi.so Asterisk Gateway Interface (AGI) 1 Running core
res_ari.so Asterisk RESTful Interface 9 Running core
res_ari_applications.so RESTful API module - Stasis application 0 Running core
res_ari_asterisk.so RESTful API module - Asterisk resources 0 Running core
res_ari_bridges.so RESTful API module - Bridge resources 0 Running core
res_ari_device_states.so RESTful API module - Device state resour 0 Running core
res_ari_endpoints.so RESTful API module - Endpoint resources 0 Running core
res_ari_events.so RESTful API module - WebSocket resource 0 Running core
res_ari_model.so ARI Model validators 9 Running core
res_ari_playbacks.so RESTful API module - Playback control re 0 Running core
res_ari_recordings.so RESTful API module - Recording resources 0 Running core
res_ari_sounds.so RESTful API module - Sound resources 0 Running core
res_audiosocket.so AudioSocket support 0 Running extended
res_clioriginate.so Call origination and redirection from th 0 Running core
res_config_curl.so Realtime Curl configuration 0 Running core
res_config_mysql.so MySQL RealTime Configuration Driver 0 Running extended
res_crypto.so Cryptographic Digital Signatures 0 Running core
res_curl.so cURL Resource Module 3 Running core
res_format_attr_h264.so H.264 Format Attribute Module 1 Running core
res_format_attr_opus.so Opus Format Attribute Module 1 Running core
res_hep.so HEPv3 API 2 Running extended
res_hep_pjsip.so PJSIP HEPv3 Logger 0 Not Running extended
res_hep_rtcp.so RTCP HEPv3 Logger 0 Not Running extended
res_http_websocket.so HTTP WebSocket Support 3 Running core
res_musiconhold.so Music On Hold Resource 0 Running core
res_phoneprov.so HTTP Phone Provisioning 1 Running extended
res_pjproject.so PJPROJECT Log and Utility Support 4 Running core
res_pjsip.so Basic SIP resource 47 Running core
res_pjsip_acl.so PJSIP ACL Resource 0 Running core
res_pjsip_authenticator_digest.so PJSIP authentication resource 0 Running core
res_pjsip_caller_id.so PJSIP Caller ID Support 1 Running core
res_pjsip_config_wizard.so PJSIP Config Wizard 1 Running core
res_pjsip_dialog_info_body_generator.so PJSIP Extension State Dialog Info+XML Pr 0 Running core
res_pjsip_diversion.so PJSIP Add Diversion Header Support 1 Running core
res_pjsip_dlg_options.so SIP OPTIONS in dialog handler 0 Running core
res_pjsip_dtmf_info.so PJSIP DTMF INFO Support 0 Running core
res_pjsip_empty_info.so PJSIP Empty INFO Support 0 Running core
res_pjsip_endpoint_identifier_anonymous.so PJSIP Anonymous endpoint identifier 0 Running core
res_pjsip_endpoint_identifier_ip.so PJSIP IP endpoint identifier 0 Running core
res_pjsip_endpoint_identifier_user.so PJSIP username endpoint identifier 0 Running core
res_pjsip_exten_state.so PJSIP Extension State Notifications 0 Running core
res_pjsip_header_funcs.so PJSIP Header Functions 0 Running core
res_pjsip_history.so PJSIP History 0 Running extended
res_pjsip_logger.so PJSIP Packet Logger 0 Running core
res_pjsip_messaging.so PJSIP Messaging Support 0 Running core
res_pjsip_mwi.so PJSIP MWI resource 0 Running core
res_pjsip_mwi_body_generator.so PJSIP MWI resource 0 Running core
res_pjsip_nat.so PJSIP NAT Support 0 Running core
res_pjsip_notify.so CLI/AMI PJSIP NOTIFY Support 0 Running core
res_pjsip_one_touch_record_info.so PJSIP INFO One Touch Recording Support 0 Running core
res_pjsip_outbound_authenticator_digest.so PJSIP authentication resource 0 Running core
res_pjsip_outbound_publish.so PJSIP Outbound Publish Support 2 Running core
res_pjsip_outbound_registration.so PJSIP Outbound Registration Support 0 Running core
res_pjsip_path.so PJSIP Path Header Support 0 Running core
res_pjsip_phoneprov_provider.so PJSIP Phoneprov Provider 0 Running extended
res_pjsip_pidf_body_generator.so PJSIP Extension State PIDF Provider 0 Running core
res_pjsip_pidf_digium_body_supplement.so PJSIP PIDF Sangoma presence supplement 0 Running core
res_pjsip_pidf_eyebeam_body_supplement.so PJSIP PIDF Eyebeam supplement 0 Running core
res_pjsip_publish_asterisk.so PJSIP Asterisk Event PUBLISH Support 0 Running core
res_pjsip_pubsub.so PJSIP event resource 12 Running core
res_pjsip_refer.so PJSIP Blind and Attended Transfer Suppor 1 Running core
res_pjsip_registrar.so PJSIP Registrar Support 0 Running core
res_pjsip_rfc3326.so PJSIP RFC3326 Support 0 Running core
res_pjsip_sdp_rtp.so PJSIP SDP RTP/AVP stream handler 0 Running core
res_pjsip_send_to_voicemail.so PJSIP REFER Send to Voicemail Support 0 Running core
res_pjsip_session.so PJSIP Session resource 19 Running core
res_pjsip_sips_contact.so UAC SIPS Contact support 0 Running core
res_pjsip_stir_shaken.so PJSIP STIR/SHAKEN Module for Asterisk 0 Running core
res_pjsip_t38.so PJSIP T.38 UDPTL Support 0 Running core
res_pjsip_transport_websocket.so PJSIP WebSocket Transport Support 0 Running core
res_pjsip_xpidf_body_generator.so PJSIP Extension State PIDF Provider 0 Running core
res_rtp_asterisk.so Asterisk RTP Stack 0 Running core
res_rtp_multicast.so Multicast RTP Engine 1 Running core
res_sorcery_astdb.so Sorcery Astdb Object Wizard 3 Running core
res_sorcery_config.so Sorcery Configuration File Object Wizard 21 Running core
res_sorcery_memory.so Sorcery In-Memory Object Wizard 8 Running core
res_sorcery_realtime.so Sorcery Realtime Object Wizard 0 Running core
res_speech.so Generic Speech Recognition API 1 Running core
res_srtp.so Secure RTP (SRTP) 0 Running core
res_stasis.so Stasis application support 13 Running core
res_stasis_device_state.so Stasis application device state support 1 Running core
res_stasis_playback.so Stasis application playback support 2 Running core
res_stasis_recording.so Stasis application recording support 3 Running core
res_stir_shaken.so STIR/SHAKEN Module for Asterisk 1 Running core
res_timing_timerfd.so Timerfd Timing Interface 0 Running core
sounds Sounds Index 1 Running core
udptl UDPTL 2 Running core
Hi Dost,
The "internal error" seems to come from openssl. Maybe some configuration issue. I don't know.
Best regards, Seb
@.*** yes its available , Thanks for this info. Can you please help understand why the DTLS failure occurs? and how can solve it ?
dmsherazi
commented
2 years ago ERROR[4104]: pjproject: <?>: icess0xb4cfa02 ...Error sending STUN request: Permission denied
is this also fatal for the webrtc call?
micmac1
commented
2 years ago ERROR[4104]: pjproject: <?>: icess0xb4cfa02 ...Error sending STUN request: Permission denied
is this also fatal for the webrtc call?
I don't know.
jslachta
commented
2 years ago Hi Dost
You need to use STUN in WebRTC for devices behind NAT.
micmac1
commented
2 years ago Closing as res_crypto is available.
I am trying to use webrtc with asterisk and I get the follwoing erros
How can i solve this , The baresip-mod-dtls-srtp package has dtls .
https://github.com/dragino/openwrt-telephony-packages/blob/master/net/asterisk-1.8.x/Makefile this Makefile has res_crypto