[Mar 29 06:27:23] VERBOSE[20008] asterisk.c: Remote UNIX connection [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'extconfig' (Configuration) [Mar 29 06:27:25] DEBUG[11548] config.c: Parsing /etc/asterisk/extconfig.conf [Mar 29 06:27:25] Asterisk 18.5.1 built by nobody @ openwrt.org on a unknown running Linux on 2022-03-10 22:32:59 UTC [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'logger' (Logger) [Mar 29 06:27:25] DEBUG[11548] config.c: Parsing /etc/asterisk/logger.conf [Mar 29 06:27:25] VERBOSE[11548] logger.c: Asterisk Queue Logger restarted [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_config_curl.so' (Realtime Curl configuration) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'cdr' (CDR Engine) [Mar 29 06:27:25] DEBUG[11548] config_options.c: cdr.conf was unchanged [Mar 29 06:27:25] NOTICE[11548] cdr.c: CDR simple logging enabled. [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'cel' (CEL Engine) [Mar 29 06:27:25] DEBUG[11548] config_options.c: cel.conf was unchanged [Mar 29 06:27:25] VERBOSE[11548] cel.c: CEL logging disabled. [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'dnsmgr' (DNS Manager) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'dsp' (DSP) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'enum' (ENUM Support) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'features' (Call Features) [Mar 29 06:27:25] DEBUG[11548] config_options.c: features.conf was unchanged [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'http' (Built-in HTTP Server) [Mar 29 06:27:25] VERBOSE[11548] tcptls.c: TLS/SSL certificate ok [Mar 29 06:27:25] DEBUG[11548] tcptls.c: Nothing changed in https server [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'indications' (Indication Tone Handling) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'acl' (Named ACL system) [Mar 29 06:27:25] DEBUG[11548] config_options.c: acl.conf was unchanged [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'manager' (Asterisk Manager Interface) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'plc' (PLC) [Mar 29 06:27:25] DEBUG[11548] config.c: Parsing /etc/asterisk/codecs.conf [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'udptl' (UDPTL) [Mar 29 06:27:25] DEBUG[11548] config_options.c: udptl.conf was unchanged [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_pjproject.so' (PJPROJECT Log and Utility Support) [Mar 29 06:27:25] DEBUG[11548] res_sorcery_config.c: Config file 'pjproject.conf' was unchanged [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_pjsip.so' (Basic SIP resource) [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[31646] res_pjsip_config_wizard.c: Config file 'pjsip_wizard.conf' was unchanged for 'auth'. [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[31646] res_pjsip_config_wizard.c: Config file 'pjsip_wizard.conf' was unchanged for 'aor'. [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[31646] res_pjsip_config_wizard.c: Config file 'pjsip_wizard.conf' was unchanged for 'endpoint'. [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[31646] res_pjsip_config_wizard.c: Config file 'pjsip_wizard.conf' was unchanged for 'identify'. [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] NOTICE[31646] sorcery.c: Type 'system' is not reloadable, maintaining previous values [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[31646] res_pjsip_config_wizard.c: Config file 'pjsip_wizard.conf' was unchanged for 'registration'. [Mar 29 06:27:25] DEBUG[31646] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] ERROR[31646] res_pjsip/config_system.c: There are no local system nameservers configured, resorting to system resolution [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_pjsip_authenticator_digest.so' (PJSIP authentication resource) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_pjsip_endpoint_identifier_ip.so' (PJSIP IP endpoint identifier) [Mar 29 06:27:25] DEBUG[11548] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[11548] res_pjsip_config_wizard.c: Config file 'pjsip_wizard.conf' was unchanged for 'identify'. [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_resolver_unbound.so' (Unbound DNS Resolver Support) [Mar 29 06:27:25] DEBUG[11548] config_options.c: resolver_unbound.conf was unchanged [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_musiconhold.so' (Music On Hold Resource) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_crypto.so' (Cryptographic Digital Signatures) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_pjsip_outbound_publish.so' (PJSIP Outbound Publish Support) [Mar 29 06:27:25] DEBUG[11548] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_rtp_asterisk.so' (Asterisk RTP Stack) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_stun_monitor.so' (STUN Network Monitor) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_pjsip_mwi.so' (PJSIP MWI resource) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_pjsip_publish_asterisk.so' (PJSIP Asterisk Event PUBLISH Support) [Mar 29 06:27:25] DEBUG[11548] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'chan_console.so' (Console Channel Driver) [Mar 29 06:27:25] DEBUG[11548] config.c: Parsing /etc/asterisk/console.conf [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'chan_dahdi.so' (DAHDI Telephony w/PRI) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_pjsip_notify.so' (CLI/AMI PJSIP NOTIFY Support) [Mar 29 06:27:25] DEBUG[11548] config_options.c: pjsip_notify.conf was unchanged [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'res_pjsip_outbound_registration.so' (PJSIP Outbound Registration Support) [Mar 29 06:27:25] DEBUG[11548] res_sorcery_config.c: Config file 'pjsip.conf' was unchanged [Mar 29 06:27:25] DEBUG[11548] res_pjsip_config_wizard.c: Config file 'pjsip_wizard.conf' was unchanged for 'registration'. [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'codec_speex.so' (Speex Coder/Decoder) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'app_confbridge.so' (Conference Bridge Application) [Mar 29 06:27:25] DEBUG[11548] config_options.c: confbridge.conf was unchanged [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'app_playback.so' (Sound File Playback Application) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'codec_dahdi.so' (Generic DAHDI Transcoder Codec Translator) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'codec_opus_open_source.so' (Opus Coder/Decoder) [Mar 29 06:27:25] VERBOSE[11548] loader.c: Reloading module 'pbx_config.so' (Text Extension Configuration) [Mar 29 06:27:25] DEBUG[11548] config.c: Parsing /etc/asterisk/extensions.conf [Mar 29 06:27:25] DEBUG[11548] pbx.c: Registered extension context 'default'; registrar: pbx_config [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '991' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '991' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '991' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '991' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '991' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '991' priority 6 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '991' priority 7 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '991' priority 8 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '991' priority 9 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '991' priority 10 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '992' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '992' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '992' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '992' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '992' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '992' priority 6 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '992' priority 7 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '992' priority 8 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '992' priority 9 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '992' priority 10 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '992' priority 11 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '999' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '999' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '999' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '999' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '999' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '998' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '998' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '998' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_90X' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_90X' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_90X' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_90X' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_90X' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_90X' priority 6 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_90X' priority 7 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_90X' priority 8 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_90X' priority 9 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_90X' priority 10 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_91X' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_91X' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_91X' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_91X' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_91X' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_91X' priority 6 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_91X' priority 7 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_91X' priority 8 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_91X' priority 9 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_91X' priority 10 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_92[0-4]' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_92[0-4]' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_92[0-4]' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_92[0-4]' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_92[0-4]' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_92[0-4]' priority 6 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_92[0-4]' priority 7 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_92[0-4]' priority 8 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_92[0-4]' priority 9 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_92[0-4]' priority 10 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_92[0-4]' priority 11 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '900' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '900' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '900' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '900' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '900' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '900' priority 6 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '900' priority 7 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '900' priority 8 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '900' priority 9 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 6 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 7 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 8 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 9 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 10 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 11 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 12 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '4000' priority 13 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_40XX' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_40XX' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_40XX' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_40XX' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_40XX' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_40XX' priority 6 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_50XX' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_50XX' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_50XX' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_50XX' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_50XX' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_50XX' priority 6 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_8XX' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_8XX' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_8XX' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_8XX' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_8XX' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_7XX' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_7XX' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_7XX' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_7XX' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_7XX' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_ZXX' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_ZXX' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '3001' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '3001' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '3001' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '3001' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '3001' priority 5 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '600' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '600' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '600' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '600' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '601' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '601' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '601' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '601' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '602' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '602' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '602' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '602' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '603' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '603' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '603' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '603' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '604' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '604' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '604' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '604' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '605' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '605' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '605' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '605' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '610' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '610' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '610' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '610' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '611' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '611' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '611' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '611' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '612' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '612' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '612' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '612' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '613' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '613' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '613' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '613' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '614' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '614' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '614' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '614' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '615' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '615' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '615' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '615' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension 'mobilephones' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension 'mobilephones' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension 'mobilephones' priority 3 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension 'mobilephones' priority 4 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension 's' priority 1 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension 's' priority 2 to default (0x213c120) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Registered extension context 'paging_handler'; registrar: pbx_config [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension 'addheader' priority 1 to paging_handler (0x2158f50) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension 'addheader' priority 2 to paging_handler (0x2158f50) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Registered extension context 'anonymous'; registrar: pbx_config [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '612' priority 1 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '612' priority 2 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '612' priority 3 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '613' priority 1 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '613' priority 2 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '613' priority 3 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '614' priority 1 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '614' priority 2 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '614' priority 3 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '615' priority 1 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '615' priority 2 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '615' priority 3 to anonymous (0x21593a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Registered extension context 'outgoing'; registrar: pbx_config [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension 's' priority 1 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension 's' priority 2 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_81XXX!' priority 1 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_81XXX!' priority 2 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_XXX!' priority 1 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_XXX!' priority 2 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_XXX!' priority 3 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_83XXX!' priority 1 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_83XXX!' priority 2 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_83XXX!' priority 3 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_84XXX!' priority 1 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_84XXX!' priority 2 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_84XXX!' priority 3 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_85XXX!' priority 1 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_85XXX!' priority 2 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Added extension '_85XXX!' priority 3 to outgoing (0x215a2a0) [Mar 29 06:27:25] DEBUG[11548] pbx.c: Registered extension context 'mobile'; registrar: pbx_config [Mar 29 06:27:25] DEBUG[11548] pbx.c: Including context 'default' in context 'mobile' [Mar 29 06:27:25] DEBUG[11548] pbx.c: Registered extension context 'fullrights'; registrar: pbx_config [Mar 29 06:27:25] DEBUG[11548] pbx.c: Including context 'default' in context 'fullrights' [Mar 29 06:27:25] DEBUG[11548] pbx.c: Including context 'outgoing' in context 'fullrights' [Mar 29 06:27:25] DEBUG[11548] pbx.c: Registered extension context 'local'; registrar: pbx_config [Mar 29 06:27:25] DEBUG[11548] pbx.c: Including context 'default' in context 'local' [Mar 29 06:27:25] DEBUG[11548] config.c: Parsing /etc/asterisk/users.conf [Mar 29 06:27:25] DEBUG[11548] pbx.c: merging incls/swits/igpats from old(local) to new(local) context, registrar = pbx_config [Mar 29 06:27:25] DEBUG[11548] pbx.c: merging incls/swits/igpats from old(fullrights) to new(fullrights) context, registrar = pbx_config [Mar 29 06:27:25] DEBUG[11548] pbx.c: merging incls/swits/igpats from old(mobile) to new(mobile) context, registrar = pbx_config [Mar 29 06:27:25] VERBOSE[11548] pbx.c: Time to scan old dialplan and merge leftovers back into the new: 0.000511 sec [Mar 29 06:27:25] VERBOSE[11548] pbx.c: Time to restore hints and swap in new dialplan: 0.000037 sec [Mar 29 06:27:25] VERBOSE[11548] pbx.c: Time to delete the old dialplan: 0.000650 sec [Mar 29 06:27:25] VERBOSE[11548] pbx.c: Total time merge_contexts_delete: 0.001198 sec [Mar 29 06:27:25] VERBOSE[11548] pbx.c: pbx_config successfully loaded 7 contexts (enable debug for details). [Mar 29 06:27:28] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (580 bytes) from TLS:188.66.164.195:48138 ---> REGISTER sip:Kindows.ddns.net:5061 SIP/2.0 Via: SIP/2.0/TLS 10.169.147.148:45546;branch=z9hG4bK380013031;rport;alias From: ;tag=1415757606 To: Call-ID: 656076758-31802-1@BJC.BGI.B.BJA CSeq: 2835 REGISTER Contact: ;reg-id=1;+sip.instance="";expires=0 Max-Forwards: 70 User-Agent: Grandstream Wave 1.0.3.34 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:28] DEBUG[31646] res_pjsip_nat.c: Saving contact '10.169.147.148:45546' [Mar 29 06:27:28] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (491 bytes) to TLS:188.66.164.195:48138 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 10.169.147.148:45546;rport=48138;received=188.66.164.195;branch=z9hG4bK380013031;alias Call-ID: 656076758-31802-1@BJC.BGI.B.BJA From: ;tag=1415757606 To: ;tag=z9hG4bK380013031 CSeq: 2835 REGISTER WWW-Authenticate: Digest realm="asterisk",nonce="1648535248/df7f5b504f2db4497a770ae992b59b9d",opaque="6017106d410cce35",algorithm=md5,qop="auth" Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:28] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (851 bytes) from TLS:188.66.164.195:48138 ---> REGISTER sip:Kindows.ddns.net:5061 SIP/2.0 Via: SIP/2.0/TLS 10.169.147.148:45546;branch=z9hG4bK449078364;rport;alias From: ;tag=1415757606 To: Call-ID: 656076758-31802-1@BJC.BGI.B.BJA CSeq: 2836 REGISTER Contact: ;reg-id=1;+sip.instance="";expires=0 Authorization: Digest username="141", realm="asterisk", nonce="1648535248/df7f5b504f2db4497a770ae992b59b9d", uri="sip:Kindows.ddns.net:5061", response="28521bb07e686e028962aa4ec3f27a67", algorithm=md5, cnonce="13857227", opaque="6017106d410cce35", qop=auth, nc=00000003 Max-Forwards: 70 User-Agent: Grandstream Wave 1.0.3.34 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:28] DEBUG[31646] res_pjsip_nat.c: Saving contact '10.169.147.148:45546' [Mar 29 06:27:28] VERBOSE[31646] res_pjsip_registrar.c: Attempted to remove non-existent contact 'sip:141@188.66.164.195:48138;transport=TLS;x-ast-orig-host=10.169.147.148:45546' from AOR '141' by request [Mar 29 06:27:28] DEBUG[31646] res_pjsip_nat.c: Restoring contact 5.37.239.43:44166 to 192.168.1.190:44166 [Mar 29 06:27:28] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (439 bytes) to TLS:188.66.164.195:48138 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 10.169.147.148:45546;rport=48138;received=188.66.164.195;branch=z9hG4bK449078364;alias Call-ID: 656076758-31802-1@BJC.BGI.B.BJA From: ;tag=1415757606 To: ;tag=z9hG4bK449078364 CSeq: 2836 REGISTER Date: Tue, 29 Mar 2022 06:27:28 GMT Contact: ;expires=3559 Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:28] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (874 bytes) from TLS:188.66.164.195:48138 ---> REGISTER sip:Kindows.ddns.net:5061 SIP/2.0 Via: SIP/2.0/TLS 10.169.147.148:45546;branch=z9hG4bK1953889568;rport;alias From: ;tag=1415757606 To: Call-ID: 656076758-31802-1@BJC.BGI.B.BJA CSeq: 2837 REGISTER Contact: ;reg-id=1;+sip.instance="" Authorization: Digest username="141", realm="asterisk", nonce="1648535248/df7f5b504f2db4497a770ae992b59b9d", uri="sip:Kindows.ddns.net:5061", response="28d4eb80c5665b60aed271b65b101bb2", algorithm=md5, cnonce="06246999", opaque="6017106d410cce35", qop=auth, nc=00000004 Max-Forwards: 70 User-Agent: Grandstream Wave 1.0.3.34 Supported: path Expires: 3600 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:28] DEBUG[31646] res_pjsip_nat.c: Saving contact '10.169.147.148:45546' [Mar 29 06:27:28] VERBOSE[31646] res_pjsip_registrar.c: Added contact 'sip:141@188.66.164.195:48138;transport=TLS;x-ast-orig-host=10.169.147.148:45546' to AOR '141' with expiration of 3600 seconds [Mar 29 06:27:28] DEBUG[11284] res_pjsip.c: 0x200b108: Wrapper created [Mar 29 06:27:28] DEBUG[11284] res_pjsip.c: 0x200b108: Set timer to 3000 msec [Mar 29 06:27:28] DEBUG[31646] res_pjsip_nat.c: Restoring contact 5.37.239.43:44166 to 192.168.1.190:44166 [Mar 29 06:27:28] DEBUG[11284] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '5.37.239.43' [Mar 29 06:27:28] DEBUG[11284] res_pjsip/pjsip_resolver.c: Transport type for target '5.37.239.43' is 'TLS transport' [Mar 29 06:27:28] DEBUG[11284] res_pjsip/pjsip_resolver.c: Target '5.37.239.43' is an IP address, skipping resolution [Mar 29 06:27:28] DEBUG[31646] res_pjsip_nat.c: Restoring contact 188.66.164.195:48138 to 10.169.147.148:45546 [Mar 29 06:27:28] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (524 bytes) to TLS:188.66.164.195:48138 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 10.169.147.148:45546;rport=48138;received=188.66.164.195;branch=z9hG4bK1953889568;alias Call-ID: 656076758-31802-1@BJC.BGI.B.BJA From: ;tag=1415757606 To: ;tag=z9hG4bK1953889568 CSeq: 2837 REGISTER Date: Tue, 29 Mar 2022 06:27:28 GMT Contact: ;expires=3559 Contact: ;expires=3599 Expires: 3600 Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:28] VERBOSE[11284] res_pjsip_logger.c: <--- Transmitting SIP request (445 bytes) to TLS:5.37.239.43:44166 ---> OPTIONS sip:141@5.37.239.43:44166;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 5.37.239.43:5061;rport;branch=z9hG4bKPjb13e30a1-d109-4e83-802a-8d5d5d39d8b7;alias From: ;tag=6ad6d8d5-a0e9-4fcb-a600-af37c4c67266 To: Contact: Call-ID: dccc9960-d482-4a8d-972e-b8429670c74e CSeq: 34300 OPTIONS Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:28] DEBUG[11284] res_pjsip.c: 0x20e9b78: Wrapper created [Mar 29 06:27:28] DEBUG[11284] res_pjsip.c: 0x20e9b78: Set timer to 3000 msec [Mar 29 06:27:28] DEBUG[11284] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '188.66.164.195' [Mar 29 06:27:28] DEBUG[11284] res_pjsip/pjsip_resolver.c: Transport type for target '188.66.164.195' is 'TLS transport' [Mar 29 06:27:28] DEBUG[11284] res_pjsip/pjsip_resolver.c: Target '188.66.164.195' is an IP address, skipping resolution [Mar 29 06:27:28] VERBOSE[11284] res_pjsip_logger.c: <--- Transmitting SIP request (451 bytes) to TLS:188.66.164.195:48138 ---> OPTIONS sip:141@188.66.164.195:48138;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 5.37.239.43:5061;rport;branch=z9hG4bKPjd0c256ba-2e70-4d1c-918f-e343e255ec23;alias From: ;tag=6bf8751e-5bac-40b2-9f17-c57662133e8e To: Contact: Call-ID: dc8e5d0d-9b91-41fa-b07b-8635faa929c1 CSeq: 58703 OPTIONS Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:28] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP response (493 bytes) from TLS:188.66.164.195:48138 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 5.37.239.43:5061;rport=5061;branch=z9hG4bKPjd0c256ba-2e70-4d1c-918f-e343e255ec23;alias From: ;tag=6bf8751e-5bac-40b2-9f17-c57662133e8e To: ;tag=665942044 Call-ID: dc8e5d0d-9b91-41fa-b07b-8635faa929c1 CSeq: 58703 OPTIONS Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.0.3.34 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:28] DEBUG[31646] res_pjsip.c: 0x20e9b78: PJSIP tsx response received [Mar 29 06:27:28] DEBUG[31646] res_pjsip.c: 0x20e9b78: Callbacks executed [Mar 29 06:27:28] DEBUG[31646] res_pjsip.c: 0x20e9b78: wrapper destroyed [Mar 29 06:27:28] VERBOSE[31646] res_pjsip/pjsip_options.c: Contact 141/sip:141@188.66.164.195:48138;transport=TLS;x-ast-orig-host=10.169.147.148:45546 is now Reachable. RTT: 52.190 msec [Mar 29 06:27:31] DEBUG[20027] res_pjsip.c: 0x200b108: Internal tsx timer expired after 3000 msec [Mar 29 06:27:31] DEBUG[20027] res_pjsip.c: 0x200b108: Callbacks executed [Mar 29 06:27:31] VERBOSE[31646] res_pjsip/pjsip_options.c: Contact 141/sip:141@5.37.239.43:44166;transport=TLS;x-ast-orig-host=192.168.1.190:44166 is now Unreachable. RTT: 0.000 msec [Mar 29 06:27:33] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (745 bytes) from UDP:192.168.1.163:5060 ---> INVITE sip:601@192.168.1.17 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK792850485 From: "F-1-1-001" ;tag=1908330358 To: Call-ID: 911058807 CSeq: 20 INVITE Contact: Content-Type: application/sdp Max-Forwards: 70 User-Agent: DnakeVoip v1.0 Content-Length: 378 v=0 o=dnake 554917775 554917775 IN IP4 5.37.239.43 s=dnake c=IN IP4 5.37.239.43 t=0 0 m=audio 6000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv m=video 6200 RTP/AVP 102 a=rtpmap:102 H264/90000 a=fmtp:102 profile-level-id=42001F; packetization-mode=1 a=ex_fmtp:102 2CIF=1 a=sendrecv [Mar 29 06:27:33] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (459 bytes) to UDP:192.168.1.163:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK792850485 Call-ID: 911058807 From: "F-1-1-001" ;tag=1908330358 To: ;tag=z9hG4bK792850485 CSeq: 20 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1648535253/a8027555ee038c5600a1763e9351492a",opaque="235b6d125d989ad2",algorithm=md5,qop="auth" Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:33] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (292 bytes) from UDP:192.168.1.163:5060 ---> ACK sip:601@192.168.1.17 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK792850485 Route: From: "F-1-1-001" ;tag=1908330358 To: ;tag=z9hG4bK792850485 Call-ID: 911058807 CSeq: 20 ACK Content-Length: 0 [Mar 29 06:27:33] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (1012 bytes) from UDP:192.168.1.163:5060 ---> INVITE sip:601@192.168.1.17 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1636673112 From: "F-1-1-001" ;tag=1908330358 To: Call-ID: 911058807 CSeq: 21 INVITE Contact: Authorization: Digest username="161", realm="asterisk", nonce="1648535253/a8027555ee038c5600a1763e9351492a", uri="sip:601@192.168.1.17", response="dcf34cd5d72e586a12d3e3f148d0aa2f", algorithm=MD5, cnonce="0a4f113b", opaque="235b6d125d989ad2", qop=auth, nc=00000001 Content-Type: application/sdp Max-Forwards: 70 User-Agent: DnakeVoip v1.0 Content-Length: 378 v=0 o=dnake 554917775 554917775 IN IP4 5.37.239.43 s=dnake c=IN IP4 5.37.239.43 t=0 0 m=audio 6000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv m=video 6200 RTP/AVP 102 a=r[Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: (null session) Request: INVITE [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Request: [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 161 [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Direct media no glare mitigation [Mar 29 06:27:33] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (287 bytes) to UDP:192.168.1.163:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1636673112 Call-ID: 911058807 From: "F-1-1-001" ;tag=1908330358 To: CSeq: 21 INVITE Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 161 Event: TSX_STATE Inv State: INCOMING [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 161 TSX State: Proceeding Inv State: INCOMING [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 161 TSX State: Proceeding Inv State: INCOMING [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Topology: Pending: (null topology) Active: (null topology) [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 161 Adding position 0 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Creating new media session [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Setting media session as default for audio [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Done [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: 161 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x20e6e78' [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) RTP allocated port 39582 [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) ICE creating session [::]:39582 (39582) [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) ICE create [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) ICE add system candidates [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) ICE add candidate: 192.168.1.17:39582, 2130706431 [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) ICE add candidate: [fd0e:aacd:db75::1]:39582, 2130706431 [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) ICE add candidate: [fe80::6:9ff:fe42:7808]:39582, 2130706431 [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) ICE add candidate: 5.37.239.43:25224, 1694498815 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: RTP instance '0x20e6e78' is setup and ready to go [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) ICE stopped [Mar 29 06:27:33] VERBOSE[31646] netsock2.c: Using SIP RTP Audio TOS bits 184 [Mar 29 06:27:33] VERBOSE[31646] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field. [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) RTCP setup on RTP instance [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: 161 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: 161 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Setting tx payload type 0 based on m type on 0xb68cd6c8 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb68cd6c8 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Crossover copying tx to rx payload mapping 0 (0x2162440) from 0xb68cd6c8 to 0xb68cd6c8 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Crossover copying tx to rx payload mapping 8 (0x2162290) from 0xb68cd6c8 to 0xb68cd6c8 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Crossover copying tx to rx payload mapping 101 (0x21622d0) from 0xb68cd6c8 to 0xb68cd6c8 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Copying rx payload mapping 0 (0x2162440) from 0xb68cd6c8 to 0x20e7034 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Copying rx payload mapping 8 (0x2162290) from 0xb68cd6c8 to 0x20e7034 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Copying rx payload mapping 101 (0x21622d0) from 0xb68cd6c8 to 0x20e7034 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 0 (0x2162440) from 0xb68cd6c8 to 0x20e7034 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 8 (0x2162290) from 0xb68cd6c8 to 0x20e7034 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 101 (0x21622d0) from 0xb68cd6c8 to 0x20e7034 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 161 Adding position 1 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Creating new media session [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Setting media session as default for video [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Done [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: 161 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x224f038' [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) RTP allocated port 36268 [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) ICE creating session [::]:36268 (36268) [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) ICE create [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) ICE add system candidates [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) ICE add candidate: 192.168.1.17:36268, 2130706431 [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) ICE add candidate: [fd0e:aacd:db75::1]:36268, 2130706431 [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) ICE add candidate: [fe80::6:9ff:fe42:7808]:36268, 2130706431 [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) ICE add candidate: 5.37.239.43:25226, 1694498815 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: RTP instance '0x224f038' is setup and ready to go [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) ICE stopped [Mar 29 06:27:33] VERBOSE[31646] netsock2.c: Using SIP RTP Video TOS bits 136 [Mar 29 06:27:33] VERBOSE[31646] netsock2.c: Using SIP RTP Video TOS bits 136 in TCLASS field. [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) RTCP setup on RTP instance [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: 161 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: 161 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Crossover copying tx to rx payload mapping 102 (0x224fe00) from 0xb68cd6c8 to 0xb68cd6c8 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Copying rx payload mapping 102 (0x224fe00) from 0xb68cd6c8 to 0x224f1f4 [Mar 29 06:27:33] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 102 (0x224fe00) from 0xb68cd6c8 to 0x224f1f4 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 161 Adding position 0 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Using existing media_session [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 161 Stream: 0:audio-0:audio:sendrecv (ulaw|alaw) [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: 161 Type: audio 0:audio-0:audio:sendrecv (ulaw|alaw) [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) RTCP ignoring duplicate property [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: RC: 1 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Had handler [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 161 Adding position 1 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Using existing media_session [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 161 Stream: 1:video-1:video:sendrecv (h264) [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: 161 Type: video 1:video-1:video:sendrecv (h264) [Mar 29 06:27:33] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) RTCP ignoring duplicate property [Mar 29 06:27:33] DEBUG[31646] res_pjsip_sdp_rtp.c: RC: 1 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Had handler [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 161 [Mar 29 06:27:33] DEBUG[31646] channel_internal_api.c: : Formats: (none) [Mar 29 06:27:33] DEBUG[31646] channel_internal_api.c: Channel is being initialized or destroyed [Mar 29 06:27:33] DEBUG[31646] stasis.c: Creating topic. name: channel:1648535253.0, detail: [Mar 29 06:27:33] DEBUG[31646] stasis.c: Topic 'channel:1648535253.0': 0x2254ca8 created [Mar 29 06:27:33] DEBUG[31646] channel.c: Channel 0x2257be8 'PJSIP/161-00000000' allocated [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> Formats: (ulaw|alaw|gsm|g726|h264|mpeg4|vp8|h263p) [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Compatible? yes [Mar 29 06:27:33] DEBUG[31646] channel_internal_api.c: PJSIP/161-00000000: MultistreamFormats: (ulaw|alaw|h264) [Mar 29 06:27:33] DEBUG[31646] channel_internal_api.c: Set native formats but not topology [Mar 29 06:27:33] DEBUG[31646] channel_internal_api.c: PJSIP/161-00000000: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] channel_internal_api.c: Used provided topology [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: PJSIP/161-00000000 [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: RC: 0 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Request: Session: PJSIP/161-00000000 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: (null session) Handled request INVITE ? yes [Mar 29 06:27:33] DEBUG[11616][C-00000001] pbx.c: Launching 'NoOp' [Mar 29 06:27:33] VERBOSE[11616][C-00000001] pbx.c: Executing [601@fullrights:1] NoOp("PJSIP/161-00000000", "EXECUTING 601 call") in new stack [Mar 29 06:27:33] DEBUG[11616][C-00000001] pbx.c: Launching 'System' [Mar 29 06:27:33] VERBOSE[11616][C-00000001] pbx.c: Executing [601@fullrights:2] System("PJSIP/161-00000000", "/usr/lib/shaula720/some_one_at_door.sh&") in new stack [Mar 29 06:27:33] DEBUG[11616][C-00000001] pbx.c: Launching 'Dial' [Mar 29 06:27:33] VERBOSE[11616][C-00000001] pbx.c: Executing [601@fullrights:3] Dial("PJSIP/161-00000000", "PJSIP/101&PJSIP/102&PJSIP/103&PJSIP/104&PJSIP/105&PJSIP/106&PJSIP/107&PJSIP/108&PJSIP/109&PJSIP/110&PJSIP/111&PJSIP/112&PJSIP/113&PJSIP/114&PJSIP/115&PJSIP/116&PJSIP/117&PJSIP/118&PJSIP/119&PJSIP/120&PJSIP/121&PJSIP/122&PJSIP/123&PJSIP/124&Local/mobilephones@default") in new stack [Mar 29 06:27:33] DEBUG[11616][C-00000001] app_dial.c: PJSIP/161-00000000: Data: PJSIP/101&PJSIP/102&PJSIP/103&PJSIP/104&PJSIP/105&PJSIP/106&PJSIP/107&PJSIP/108&PJSIP/109&PJSIP/110&PJSIP/111&PJSIP/112&PJSIP/113&PJSIP/114&PJSIP/115&PJSIP/116&PJSIP/117&PJSIP/118&PJSIP/119&PJSIP/120&PJSIP/121&PJSIP/122&PJSIP/123&PJSIP/124&Local/mobilephones@default [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 101 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 101 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 101 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '101': Could not create dialog to invalid URI '101'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '101' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 102 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 102 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 102 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '102': Could not create dialog to invalid URI '102'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '102' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 103 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 103 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 103 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '103': Could not create dialog to invalid URI '103'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '103' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 104 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 104 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 104 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '104': Could not create dialog to invalid URI '104'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '104' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 105 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 105 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 105 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '105': Could not create dialog to invalid URI '105'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '105' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 106 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 106 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 106 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '106': Could not create dialog to invalid URI '106'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '106' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 107 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 107 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 107 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '107': Could not create dialog to invalid URI '107'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '107' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 108 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 108 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 108 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '108': Could not create dialog to invalid URI '108'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '108' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 109 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 109 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 109 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '109': Could not create dialog to invalid URI '109'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '109' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 110 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 110 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 110 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '110': Could not create dialog to invalid URI '110'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '110' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 111 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 111 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 111 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '111': Could not create dialog to invalid URI '111'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '111' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 112 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 112 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 112 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '112': Could not create dialog to invalid URI '112'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '112' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 113 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 113 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 113 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '113': Could not create dialog to invalid URI '113'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '113' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 114 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 114 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 114 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '114': Could not create dialog to invalid URI '114'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '114' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 115 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 115 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 115 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '115': Could not create dialog to invalid URI '115'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '115' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 116 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 116 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 116 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '116': Could not create dialog to invalid URI '116'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '116' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 117 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 117 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 117 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '117': Could not create dialog to invalid URI '117'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '117' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 118 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 118 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 118 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '118': Could not create dialog to invalid URI '118'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '118' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 119 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 119 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 119 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '119': Could not create dialog to invalid URI '119'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '119' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 120 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 120 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 120 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '120': Could not create dialog to invalid URI '120'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '120' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 121 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 121 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 121 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '121': Could not create dialog to invalid URI '121'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '121' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 122 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 122 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 122 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '122': Could not create dialog to invalid URI '122'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '122' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 123 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 123 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 123 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '123': Could not create dialog to invalid URI '123'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '123' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: 124 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: 124 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: 124 (null) Topology: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] ERROR[31646] res_pjsip.c: Endpoint '124': Could not create dialog to invalid URI '124'. Is endpoint registered and reachable? [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Couldn't create dialog [Mar 29 06:27:33] ERROR[31646] chan_pjsip.c: Failed to create outgoing session to endpoint '124' [Mar 29 06:27:33] DEBUG[31646] chan_pjsip.c: Couldn't create session [Mar 29 06:27:33] DEBUG[11616][C-00000001] chan_pjsip.c: Couldn't push task [Mar 29 06:27:33] WARNING[11616][C-00000001] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: : Formats: (none) [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: Channel is being initialized or destroyed [Mar 29 06:27:33] DEBUG[11616][C-00000001] stasis.c: Creating topic. name: channel:1648535253.1, detail: [Mar 29 06:27:33] DEBUG[11616][C-00000001] stasis.c: Topic 'channel:1648535253.1': 0x221de98 created [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel.c: Channel 0x2221348 'Local/mobilephones@default-00000000;1' allocated [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: Local/mobilephones@default-00000000;1: MultistreamFormats: (ulaw|h264) [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: Set native formats but not topology [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: Local/mobilephones@default-00000000;1: <0:audio-0:audio:sendrecv (ulaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: Used provided topology [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: : Formats: (none) [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: Channel is being initialized or destroyed [Mar 29 06:27:33] DEBUG[11616][C-00000001] stasis.c: Creating topic. name: channel:1648535253.2, detail: [Mar 29 06:27:33] DEBUG[11616][C-00000001] stasis.c: Topic 'channel:1648535253.2': 0x2132078 created [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel.c: Channel 0x2224cf8 'Local/mobilephones@default-00000000;2' allocated [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: Local/mobilephones@default-00000000;2: MultistreamFormats: (ulaw|h264) [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: Set native formats but not topology [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: Local/mobilephones@default-00000000;2: <0:audio-0:audio:sendrecv (ulaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel_internal_api.c: Used provided topology [Mar 29 06:27:33] VERBOSE[11616][C-00000001] app_dial.c: Called Local/mobilephones@default [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;1 setting read format path: ulaw -> ulaw [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel.c: Channel PJSIP/161-00000000 setting write format path: ulaw -> ulaw [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel.c: Channel PJSIP/161-00000000 setting read format path: ulaw -> ulaw [Mar 29 06:27:33] DEBUG[11616][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;1 setting write format path: ulaw -> ulaw [Mar 29 06:27:33] DEBUG[11620][C-00000001] pbx.c: Launching 'Ringing' [Mar 29 06:27:33] VERBOSE[11620][C-00000001] pbx.c: Executing [mobilephones@default:1] Ringing("Local/mobilephones@default-00000000;2", "") in new stack [Mar 29 06:27:33] VERBOSE[11616][C-00000001] app_dial.c: Local/mobilephones@default-00000000;1 is ringing [Mar 29 06:27:33] DEBUG[11620][C-00000001] pbx.c: Launching 'System' [Mar 29 06:27:33] VERBOSE[11620][C-00000001] pbx.c: Executing [mobilephones@default:2] System("Local/mobilephones@default-00000000;2", "/bin/sleep 6") in new stack [Mar 29 06:27:33] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (474 bytes) to UDP:192.168.1.163:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1636673112 Call-ID: 911058807 From: "F-1-1-001" ;tag=1908330358 To: ;tag=435dfe33-def3-4ea2-bd8e-e8f5b062044f CSeq: 21 INVITE Server: SHAULA-001(7.4.0) Contact: Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Content-Length: 0 [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 Event: TSX_STATE Inv State: EARLY [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Proceeding Inv State: EARLY [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Proceeding Inv State: EARLY [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: Topology: Pending: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> Active: (null topology) [Mar 29 06:27:33] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:36] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (301 bytes) from UDP:192.168.1.163:5060 ---> OPTIONS sip:192.168.1.17 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1252843215 From: ;tag=1821195513 To: Call-ID: 382961508 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 70 User-Agent: DnakeVoip v1.0 Content-Length: 0 [Mar 29 06:27:36] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (446 bytes) to UDP:192.168.1.163:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1252843215 Call-ID: 382961508 From: ;tag=1821195513 To: ;tag=z9hG4bK1252843215 CSeq: 20 OPTIONS WWW-Authenticate: Digest realm="asterisk",nonce="1648535256/52ab80c9cf80999f7697d2bc53e2bb6d",opaque="5a522b616311da9f",algorithm=md5,qop="auth" Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:36] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (562 bytes) from UDP:192.168.1.163:5060 ---> OPTIONS sip:192.168.1.17 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK236589572 From: ;tag=1821195513 To: Call-ID: 382961508 CSeq: 21 OPTIONS Authorization: Digest username="161", realm="asterisk", nonce="1648535256/52ab80c9cf80999f7697d2bc53e2bb6d", uri="sip:192.168.1.17", response="6472d80b6ea1e22ad1403cd3db40092c", algorithm=MD5, cnonce="0a4f113b", opaque="5a522b616311da9f", qop=auth, nc=00000001 Accept: application/sdp Max-Forwards: 70 User-Agent: DnakeVoip v1.0 Content-Length: 0 [Mar 29 06:27:36] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (773 bytes) to UDP:192.168.1.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK236589572 Call-ID: 382961508 From: ;tag=1821195513 To: ;tag=z9hG4bK236589572 CSeq: 21 OPTIONS Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Accept-Encoding: identity Accept-Language: en Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:39] DEBUG[11620][C-00000001] pbx_variables.c: Function PJSIP_DIAL_CONTACTS(140) result is '' [Mar 29 06:27:39] DEBUG[11620][C-00000001] pbx_variables.c: Function PJSIP_DIAL_CONTACTS(141) result is 'PJSIP/141/sip:141@188.66.164.195:48138;transport=TLS;x-ast-orig-host=10.169.147.148:45546' [Mar 29 06:27:39] DEBUG[11620][C-00000001] pbx.c: Launching 'Dial' [Mar 29 06:27:39] VERBOSE[11620][C-00000001] pbx.c: Executing [mobilephones@default:3] Dial("Local/mobilephones@default-00000000;2", "&PJSIP/141/sip:141@188.66.164.195:48138;transport=TLS;x-ast-orig-host=10.169.147.148:45546") in new stack [Mar 29 06:27:39] DEBUG[11620][C-00000001] app_dial.c: Local/mobilephones@default-00000000;2: Data: &PJSIP/141/sip:141@188.66.164.195:48138;transport=TLS;x-ast-orig-host=10.169.147.148:45546 [Mar 29 06:27:39] DEBUG[11620][C-00000001] chan_pjsip.c: 141/sip:141@188.66.164.195:48138;transport=TLS;x-ast-orig-host=10.169.147.148:45546 Topology: <0:audio-0:audio:sendrecv (ulaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:39] DEBUG[31646] chan_pjsip.c: 141/sip:141@188.66.164.195:48138;transport=TLS;x-ast-orig-host=10.169.147.148:45546 [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: 141 (null) Topology: <0:audio-0:audio:sendrecv (ulaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:39] DEBUG[31646] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Mar 29 06:27:39] DEBUG[31646] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Mar 29 06:27:39] DEBUG[31646] chan_pjsip.c: 141 [Mar 29 06:27:39] DEBUG[31646] chan_pjsip.c: Direct media no glare mitigation [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:39] DEBUG[31646] chan_pjsip.c: [Mar 29 06:27:39] DEBUG[11620][C-00000001] chan_pjsip.c: 141 [Mar 29 06:27:39] DEBUG[11620][C-00000001] channel_internal_api.c: : Formats: (none) [Mar 29 06:27:39] DEBUG[11620][C-00000001] channel_internal_api.c: Channel is being initialized or destroyed [Mar 29 06:27:39] DEBUG[11620][C-00000001] stasis.c: Creating topic. name: channel:1648535259.3, detail: [Mar 29 06:27:39] DEBUG[11620][C-00000001] stasis.c: Topic 'channel:1648535259.3': 0x2134688 created [Mar 29 06:27:39] DEBUG[11620][C-00000001] channel.c: Channel 0x2134a98 'PJSIP/141-00000001' allocated [Mar 29 06:27:39] DEBUG[11620][C-00000001] chan_pjsip.c: Topology: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p)> Formats: (ulaw|alaw|gsm|g726|h264|mpeg4|vp8|h263p) [Mar 29 06:27:39] DEBUG[11620][C-00000001] chan_pjsip.c: Compatible? yes [Mar 29 06:27:39] DEBUG[11620][C-00000001] channel_internal_api.c: PJSIP/141-00000001: MultistreamFormats: (ulaw|alaw|gsm|g726|h264|mpeg4|vp8|h263p) [Mar 29 06:27:39] DEBUG[11620][C-00000001] channel_internal_api.c: Set native formats but not topology [Mar 29 06:27:39] DEBUG[11620][C-00000001] channel_internal_api.c: PJSIP/141-00000001: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p)> [Mar 29 06:27:39] DEBUG[11620][C-00000001] channel_internal_api.c: Used provided topology [Mar 29 06:27:39] DEBUG[11620][C-00000001] chan_pjsip.c: [Mar 29 06:27:39] DEBUG[11620][C-00000001] chan_pjsip.c: Channel: PJSIP/141-00000001 [Mar 29 06:27:39] DEBUG[11620][C-00000001] rtp_engine.c: Can't find native functions for channel 'Local/mobilephones@default-00000000;2' [Mar 29 06:27:39] DEBUG[11620][C-00000001] chan_pjsip.c: PJSIP/141-00000001 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p)> [Mar 29 06:27:39] DEBUG[11620][C-00000001] chan_pjsip.c: 'call' task pushed [Mar 29 06:27:39] DEBUG[31646] chan_pjsip.c: PJSIP/141-00000001 Topology: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p)> [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Adding position 0 [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: Creating new media session [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: Setting media session as default for audio [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: Done [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Stream: 0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726) [Mar 29 06:27:39] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/141-00000001 Type: audio 0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726) [Mar 29 06:27:39] DEBUG[31646] res_pjsip_sdp_rtp.c: Transport transport-tls bound to 0.0.0.0: Using it for RTP media. [Mar 29 06:27:39] DEBUG[31646] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2148e58' [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) RTP allocated port 35956 [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE creating session 0.0.0.0:35956 (35956) [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE create [Mar 29 06:27:39] VERBOSE[11620][C-00000001] app_dial.c: Called PJSIP/141/sip:141@188.66.164.195:48138;transport=TLS;x-ast-orig-host=10.169.147.148:45546 [Mar 29 06:27:39] DEBUG[11620][C-00000001] channel.c: Channel PJSIP/141-00000001 setting read format path: ulaw -> ulaw [Mar 29 06:27:39] DEBUG[11620][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;2 setting write format path: ulaw -> ulaw [Mar 29 06:27:39] DEBUG[11620][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;2 setting read format path: ulaw -> ulaw [Mar 29 06:27:39] DEBUG[11620][C-00000001] channel.c: Channel PJSIP/141-00000001 setting write format path: ulaw -> ulaw [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE add system candidates [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE add candidate: 192.168.1.17:35956, 2130706431 [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE add candidate: 5.37.239.43:25240, 1694498815 [Mar 29 06:27:39] DEBUG[31646] rtp_engine.c: RTP instance '0x2148e58' is setup and ready to go [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE change number of components 2 -> 1 [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) RTCP setup on RTP instance [Mar 29 06:27:39] DEBUG[31646] res_srtp.c: local_key64 uyxcmCOrIWBYx2cxgk+KMlSGvRyGVhhSTOJN8tOn len 40 [Mar 29 06:27:39] DEBUG[31646] res_pjsip_sdp_rtp.c: RC: 1 [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: Handled [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Adding position 1 [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: Creating new media session [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: Setting media session as default for video [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: Done [Mar 29 06:27:39] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Stream: 1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p) [Mar 29 06:27:39] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/141-00000001 Type: video 1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p) [Mar 29 06:27:39] DEBUG[31646] res_pjsip_sdp_rtp.c: Transport transport-tls bound to 0.0.0.0: Using it for RTP media. [Mar 29 06:27:39] DEBUG[31646] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x214a688' [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) RTP allocated port 35492 [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) ICE creating session 0.0.0.0:35492 (35492) [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) ICE create [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) ICE add system candidates [Mar 29 06:27:39] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) ICE add candidate: 192.168.1.17:35492, 2130706431 [Mar 29 06:27:40] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) ICE add candidate: 5.37.239.43:25244, 1694498815 [Mar 29 06:27:40] DEBUG[31646] rtp_engine.c: RTP instance '0x214a688' is setup and ready to go [Mar 29 06:27:40] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) RTCP setup on RTP instance [Mar 29 06:27:40] DEBUG[31646] res_pjsip_sdp_rtp.c: RC: 1 [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: Handled [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:40] DEBUG[11620][C-00000001] channel.c: Local/mobilephones@default-00000000;2: Dropping redundant connected line update "141" . [Mar 29 06:27:40] DEBUG[31646] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '188.66.164.195' [Mar 29 06:27:40] DEBUG[31646] res_pjsip/pjsip_resolver.c: Transport type for target '188.66.164.195' is 'TLS transport' [Mar 29 06:27:40] DEBUG[31646] res_pjsip/pjsip_resolver.c: Target '188.66.164.195' is an IP address, skipping resolution [Mar 29 06:27:40] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP request (2491 bytes) to TLS:188.66.164.195:48138 ---> INVITE sip:141@188.66.164.195:48138;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 5.37.239.43:5061;rport;branch=z9hG4bKPj3a6d7111-10cc-4749-bfe0-56785b179871;alias From: "161" ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c To: Contact: Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 CSeq: 2242 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "161" Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Type: application/sdp Content-Length: 1737 v=0 o=- 658550646 658550646 IN IP4 5.37.239.43 s=Asterisk c=IN IP4 5.37.239.43 t=0 0 a=msid-semantic:WMS * a=group:BUNDLE audio-0 vide[Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Event: TSX_STATE Inv State: CALLING [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Calling Inv State: CALLING [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Calling Inv State: CALLING [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: Topology: Pending: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p)> Active: (null topology) [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:40] DEBUG[31646] chan_pjsip.c: RC: 0 [Mar 29 06:27:40] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP response (490 bytes) from TLS:188.66.164.195:48138 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 5.37.239.43:5061;rport=5061;branch=z9hG4bKPj3a6d7111-10cc-4749-bfe0-56785b179871;alias From: "161" ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c To: Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 CSeq: 2242 INVITE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.0.3.34 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Method: INVITE Status: 100 [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Proceeding Inv State: CALLING [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Proceeding Inv State: CALLING [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: Topology: Pending: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p)> Active: (null topology) [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:40] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP response (622 bytes) from TLS:188.66.164.195:48138 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 5.37.239.43:5061;rport=5061;branch=z9hG4bKPj3a6d7111-10cc-4749-bfe0-56785b179871;alias From: "161" ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c To: ;tag=1994111823 Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 CSeq: 2242 INVITE Contact: Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.0.3.34 Allow-Events: talk, hold Require: 100rel RSeq: 100 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Method: INVITE Status: 180 [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Event: TSX_STATE Inv State: EARLY [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:40] DEBUG[31646] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '188.66.164.195' [Mar 29 06:27:40] DEBUG[31646] res_pjsip/pjsip_resolver.c: Transport type for target '188.66.164.195' is 'TLS transport' [Mar 29 06:27:40] DEBUG[31646] res_pjsip/pjsip_resolver.c: Target '188.66.164.195' is an IP address, skipping resolution [Mar 29 06:27:40] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP request (442 bytes) to TLS:188.66.164.195:48138 ---> PRACK sip:141@188.66.164.195:48138;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 5.37.239.43:5061;rport;branch=z9hG4bKPj04d602a5-9748-4e5c-b92f-e6b68b754a09;alias From: "161" ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c To: ;tag=1994111823 Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 CSeq: 2243 PRACK RAck: 100 2242 INVITE Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Calling Inv State: EARLY [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Calling Inv State: EARLY [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: Topology: Pending: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p)> Active: (null topology) [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Proceeding Inv State: EARLY [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Proceeding Inv State: EARLY [Mar 29 06:27:40] VERBOSE[11620][C-00000001] app_dial.c: PJSIP/141-00000001 is ringing [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: Topology: Pending: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p)> Active: (null topology) [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:40] VERBOSE[11616][C-00000001] app_dial.c: Local/mobilephones@default-00000000;1 is ringing [Mar 29 06:27:40] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP response (562 bytes) from TLS:188.66.164.195:48138 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 5.37.239.43:5061;rport=5061;branch=z9hG4bKPj04d602a5-9748-4e5c-b92f-e6b68b754a09;alias From: "161" ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c To: ;tag=1994111823 Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 CSeq: 2243 PRACK Contact: Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.0.3.34 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Method: PRACK Status: 200 [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Completed Inv State: EARLY [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001: PRACK received final response code 200 [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Completed Inv State: EARLY [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: Topology: Pending: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p)> Active: (null topology) [Mar 29 06:27:40] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:40] DEBUG[20027] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Terminated Inv State: EARLY [Mar 29 06:27:40] DEBUG[20027] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:40] DEBUG[20027] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Terminated Inv State: EARLY [Mar 29 06:27:40] DEBUG[20027] res_pjsip_session.c: Topology: Pending: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p)> Active: (null topology) [Mar 29 06:27:40] DEBUG[20027] res_pjsip_session.c: [Mar 29 06:27:40] DEBUG[11281] threadpool.c: Worker thread idle timeout reached. Dying. [Mar 29 06:27:40] DEBUG[11288] threadpool.c: Worker thread idle timeout reached. Dying. [Mar 29 06:27:40] DEBUG[11283] threadpool.c: Worker thread idle timeout reached. Dying. [Mar 29 06:27:40] DEBUG[11286] threadpool.c: Worker thread idle timeout reached. Dying. [Mar 29 06:27:40] DEBUG[11282] threadpool.c: Worker thread idle timeout reached. Dying. [Mar 29 06:27:40] DEBUG[11287] threadpool.c: Worker thread idle timeout reached. Dying. [Mar 29 06:27:40] DEBUG[11280] threadpool.c: Worker thread idle timeout reached. Dying. [Mar 29 06:27:43] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP response (1100 bytes) from TLS:188.66.164.195:48138 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 5.37.239.43:5061;rport=5061;branch=z9hG4bKPj3a6d7111-10cc-4749-bfe0-56785b179871;alias From: "161" ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c To: ;tag=1994111823 Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 CSeq: 2242 INVITE Contact: Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.0.3.34 Session-Expires: 1800;refresher=uac Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 449 v=0 o=141 8000 8000 IN IP4 10.169.147.148 s=SIP Call c=IN IP4 10.169.147.148 t=0 0 m=audio 26892 RTP/AVP 0 8 3 111 a=sendrecv a=rtcp:26893 IN IP4 10.169.147.148 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000[Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Method: INVITE Status: 200 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Event: TSX_STATE Inv State: CONNECTING [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/141-00000001 Stream: 0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726) [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE change number of components 1 -> 2 [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE add system candidates [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE add candidate: 192.168.1.17:35957, 2130706430 [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE add candidate: 5.37.239.43:25255, 1694498814 [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) RTCP setup on RTP instance [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) RTCP setting address on RTP instance [Mar 29 06:27:43] VERBOSE[31646] res_rtp_asterisk.c: 0x221e0c0 -- Strict RTP learning after remote address set to: 10.169.147.148:26892 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: (0x2148e58) ICE process attributes [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: (0x2148e58) ICE no, or invalid ice-ufrag [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/141-00000001 ANSWER [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/141-00000001 [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Setting tx payload type 0 based on m type on 0xb68cd520 [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb68cd520 [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Setting tx payload type 3 based on m type on 0xb68cd520 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 0 (0x1f83ef0) from 0xb68cd520 to 0x2149014 [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 3 (0x2115330) from 0xb68cd520 to 0x2149014 [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 8 (0x21152f0) from 0xb68cd520 to 0x2149014 [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 111 (0x222d7d0) from 0xb68cd520 to 0x2149014 [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: PJSIP/141-00000001: MultistreamFormats: (h264|mpeg4|vp8|h263p|ulaw) [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: Set native formats but not topology [Mar 29 06:27:43] DEBUG[31646] channel.c: Channel PJSIP/141-00000001 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[31646] channel.c: Channel PJSIP/141-00000001 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: Handled [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001: Applied negotiated SDP media stream 'audio' using audio SDP handler [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/141-00000001 Stream: 1:video-1:video:sendrecv (h264|mpeg4|vp8|h263p) [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) ICE add system candidates [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) ICE add candidate: 192.168.1.17:35493, 2130706430 [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) ICE add candidate: 5.37.239.43:25259, 1694498814 [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) RTCP setup on RTP instance [Mar 29 06:27:43] DEBUG[31646] res_srtp.c: local_key64 bJBSHt4uGUgx1LPKSsNqOs32PwC/FumOIu/9AXm0 len 40 [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) RTCP setting address on RTP instance [Mar 29 06:27:43] VERBOSE[31646] res_rtp_asterisk.c: 0x2226780 -- Strict RTP learning after remote address set to: 10.169.147.148:19774 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: (0x214a688) ICE process attributes [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: (0x214a688) ICE no, or invalid ice-ufrag [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/141-00000001 ANSWER [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/141-00000001 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 99 (0x2247860) from 0xb68cd520 to 0x214a844 [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: PJSIP/141-00000001: MultistreamFormats: (ulaw|h264) [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: Set native formats but not topology [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: moh [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001: Applied negotiated SDP media stream 'video' using video SDP handler [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: PJSIP/141-00000001: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: Used provided topology [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:43] DEBUG[31646] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '188.66.164.195' [Mar 29 06:27:43] DEBUG[31646] res_pjsip/pjsip_resolver.c: Transport type for target '188.66.164.195' is 'TLS transport' [Mar 29 06:27:43] DEBUG[31646] res_pjsip/pjsip_resolver.c: Target '188.66.164.195' is an IP address, skipping resolution [Mar 29 06:27:43] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP request (415 bytes) to TLS:188.66.164.195:48138 ---> ACK sip:141@188.66.164.195:48138;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 5.37.239.43:5061;rport;branch=z9hG4bKPj6ea03932-a53e-4923-b66f-33c8fd66ce34;alias From: "161" ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c To: ;tag=1994111823 Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 CSeq: 2242 ACK Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Event: TX_MSG Inv State: CONFIRMED [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Terminated Inv State: CONFIRMED [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Terminated Inv State: CONFIRMED [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: Topology: Pending: (null topology) Active: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:43] VERBOSE[11620][C-00000001] app_dial.c: PJSIP/141-00000001 answered Local/mobilephones@default-00000000;2 [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Channel PJSIP/141-00000001 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;2 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;2 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Channel PJSIP/141-00000001 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11620][C-00000001] stasis.c: Creating topic. name: bridge:all/bridge:c806adff-82ca-42f1-8093-5ff2be7e3cb1, detail: [Mar 29 06:27:43] DEBUG[11620][C-00000001] stasis.c: Topic 'bridge:all/bridge:c806adff-82ca-42f1-8093-5ff2be7e3cb1': 0x2226108 created [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge_native_rtp.c: Bridge 'c806adff-82ca-42f1-8093-5ff2be7e3cb1' can not use native RTP bridge as two channels are required [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11620][C-00000001] dahdi/bridge_native_dahdi.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: Cannot use native DAHDI. Must have two channels. [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Chose bridge technology simple_bridge [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: calling simple_bridge technology constructor [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: calling simple_bridge technology start [Mar 29 06:27:43] DEBUG[11681][C-00000001] bridge_channel.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: 0x224a198(PJSIP/141-00000001) is joining [Mar 29 06:27:43] DEBUG[11681][C-00000001] bridge_channel.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: pushing 0x224a198(PJSIP/141-00000001) [Mar 29 06:27:43] VERBOSE[11616][C-00000001] app_dial.c: Local/mobilephones@default-00000000;1 answered PJSIP/161-00000000 [Mar 29 06:27:43] VERBOSE[11681][C-00000001] bridge_channel.c: Channel PJSIP/141-00000001 joined 'simple_bridge' basic-bridge [Mar 29 06:27:43] DEBUG[11681][C-00000001] bridge_native_rtp.c: Bridge 'c806adff-82ca-42f1-8093-5ff2be7e3cb1' can not use native RTP bridge as two channels are required [Mar 29 06:27:43] DEBUG[11681][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11681][C-00000001] dahdi/bridge_native_dahdi.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: Cannot use native DAHDI. Must have two channels. [Mar 29 06:27:43] DEBUG[11681][C-00000001] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11681][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11681][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11681][C-00000001] bridge.c: Chose bridge technology simple_bridge [Mar 29 06:27:43] DEBUG[11681][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1 is already using the new technology. [Mar 29 06:27:43] DEBUG[11681][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: 0x224a198(PJSIP/141-00000001) is joining simple_bridge technology [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge_channel.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: 0x2115108(Local/mobilephones@default-00000000;2) is joining [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge_channel.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: pushing 0x2115108(Local/mobilephones@default-00000000;2) [Mar 29 06:27:43] VERBOSE[11620][C-00000001] bridge_channel.c: Channel Local/mobilephones@default-00000000;2 joined 'simple_bridge' basic-bridge [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge_native_rtp.c: Bridge 'c806adff-82ca-42f1-8093-5ff2be7e3cb1'. Checking compatability for channels 'PJSIP/141-00000001' and 'Local/mobilephones@default-00000000;2' [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge_native_rtp.c: Bridge 'c806adff-82ca-42f1-8093-5ff2be7e3cb1' can not use native RTP bridge as could not get details [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11620][C-00000001] dahdi/bridge_native_dahdi.c: Channel 'PJSIP/141-00000001' is not DAHDI. [Mar 29 06:27:43] DEBUG[11620][C-00000001] dahdi/bridge_native_dahdi.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: Cannot use native DAHDI. Channel 'PJSIP/141-00000001' not compatible. [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[20021] cdr.c: Finalized CDR for PJSIP/141-00000001 - start 1648535259.776966 answer 1648535263.726658 end 1648535263.729693 dur 3.952 bill 0.003 dispo ANSWERED [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Chose bridge technology simple_bridge [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1 is already using the new technology. [Mar 29 06:27:43] DEBUG[11620][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: 0x2115108(Local/mobilephones@default-00000000;2) is joining simple_bridge technology [Mar 29 06:27:43] DEBUG[11616][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;1 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11616][C-00000001] channel.c: Channel PJSIP/161-00000000 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11616][C-00000001] channel.c: Channel PJSIP/161-00000000 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11616][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;1 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11616][C-00000001] chan_pjsip.c: PJSIP/161-00000000 [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;2 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Channel PJSIP/141-00000001 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Channel PJSIP/141-00000001 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;2 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Local/mobilephones@default-00000000;2: Topologies already match. Current: <0:audio-0:audio:sendrecv (ulaw)> <1:video-1:video:sendrecv (h264)> Requested: <0:audio-0:audio:sendrecv (ulaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:43] DEBUG[31646] chan_pjsip.c: PJSIP/161-00000000 [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: PJSIP/141-00000001: Topologies already match. Current: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264)> Requested: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/161-00000000 Stream: 0:audio-0:audio:sendrecv (ulaw|alaw) [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) RTCP ignoring duplicate property [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x20e6e78) RTCP setting address on RTP instance [Mar 29 06:27:43] VERBOSE[31646] res_rtp_asterisk.c: 0x2250760 -- Strict RTP learning after remote address set to: 5.37.239.43:6000 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/161-00000000 ANSWER [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/161-00000000 [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Setting tx payload type 0 based on m type on 0xb68cd5e8 [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb68cd5e8 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 0 (0x212f350) from 0xb68cd5e8 to 0x20e7034 [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 8 (0x224ebb0) from 0xb68cd5e8 to 0x20e7034 [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 101 (0x22479f0) from 0xb68cd5e8 to 0x20e7034 [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: PJSIP/161-00000000: MultistreamFormats: (h264|ulaw) [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: Set native formats but not topology [Mar 29 06:27:43] DEBUG[31646] channel.c: Channel PJSIP/161-00000000 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[31646] channel.c: Channel PJSIP/161-00000000 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: Handled [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000: Applied negotiated SDP media stream 'audio' using audio SDP handler [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/161-00000000 Stream: 1:video-1:video:sendrecv (h264) [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) RTCP ignoring duplicate property [Mar 29 06:27:43] DEBUG[31646] res_rtp_asterisk.c: (0x224f038) RTCP setting address on RTP instance [Mar 29 06:27:43] VERBOSE[31646] res_rtp_asterisk.c: 0x212c080 -- Strict RTP learning after remote address set to: 5.37.239.43:6200 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/161-00000000 ANSWER [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: PJSIP/161-00000000 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:43] DEBUG[31646] rtp_engine.c: Copying tx payload mapping 102 (0x22492c0) from 0xb68cd5f0 to 0x224f1f4 [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: PJSIP/161-00000000: MultistreamFormats: (ulaw|h264) [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: Set native formats but not topology [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: [Mar 29 06:27:43] DEBUG[31646] res_pjsip_sdp_rtp.c: moh [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000: Applied negotiated SDP media stream 'video' using video SDP handler [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: PJSIP/161-00000000: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:43] DEBUG[31646] channel_internal_api.c: Used provided topology [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:43] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (993 bytes) to UDP:192.168.1.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1636673112 Call-ID: 911058807 From: "F-1-1-001" ;tag=1908330358 To: ;tag=435dfe33-def3-4ea2-bd8e-e8f5b062044f CSeq: 21 INVITE Server: SHAULA-001(7.4.0) Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Contact: Supported: 100rel, timer, replaces, norefersub P-Asserted-Identity: "141" Content-Type: application/sdp Content-Length: 380 v=0 o=- 554917775 554917777 IN IP4 192.168.1.17 s=Asterisk c=IN IP4 192.168.1.17 t=0 0 m=audio 39582 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv m=video 36268 RTP/AVP 10[Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 Event: TSX_STATE Inv State: CONNECTING [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Completed Inv State: CONNECTING [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Completed Inv State: CONNECTING [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: Topology: Pending: (null topology) Active: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:43] DEBUG[31646] chan_pjsip.c: [Mar 29 06:27:43] DEBUG[11616][C-00000001] chan_pjsip.c: [Mar 29 06:27:43] DEBUG[11616][C-00000001] stasis.c: Creating topic. name: bridge:all/bridge:fd2791bb-3e58-4978-9bc6-850dacedf3f7, detail: [Mar 29 06:27:43] DEBUG[11616][C-00000001] stasis.c: Topic 'bridge:all/bridge:fd2791bb-3e58-4978-9bc6-850dacedf3f7': 0x2255888 created [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge_native_rtp.c: Bridge 'fd2791bb-3e58-4978-9bc6-850dacedf3f7' can not use native RTP bridge as two channels are required [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11616][C-00000001] dahdi/bridge_native_dahdi.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: Cannot use native DAHDI. Must have two channels. [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Chose bridge technology simple_bridge [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: calling simple_bridge technology constructor [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: calling simple_bridge technology start [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_channel.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: 0x2246c98(Local/mobilephones@default-00000000;1) is joining [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_channel.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: pushing 0x2246c98(Local/mobilephones@default-00000000;1) [Mar 29 06:27:43] VERBOSE[11683][C-00000001] bridge_channel.c: Channel Local/mobilephones@default-00000000;1 joined 'simple_bridge' basic-bridge [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_native_rtp.c: Bridge 'fd2791bb-3e58-4978-9bc6-850dacedf3f7' can not use native RTP bridge as two channels are required [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11683][C-00000001] dahdi/bridge_native_dahdi.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: Cannot use native DAHDI. Must have two channels. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Chose bridge technology simple_bridge [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7 is already using the new technology. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: 0x2246c98(Local/mobilephones@default-00000000;1) is joining simple_bridge technology [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge_channel.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: 0x2114d98(PJSIP/161-00000000) is joining [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge_channel.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: pushing 0x2114d98(PJSIP/161-00000000) [Mar 29 06:27:43] VERBOSE[11616][C-00000001] bridge_channel.c: Channel PJSIP/161-00000000 joined 'simple_bridge' basic-bridge [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge_native_rtp.c: Bridge 'fd2791bb-3e58-4978-9bc6-850dacedf3f7'. Checking compatability for channels 'Local/mobilephones@default-00000000;1' and 'PJSIP/161-00000000' [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge_native_rtp.c: Bridge 'fd2791bb-3e58-4978-9bc6-850dacedf3f7' can not use native RTP bridge as could not get details [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11616][C-00000001] dahdi/bridge_native_dahdi.c: Channel 'Local/mobilephones@default-00000000;1' is not DAHDI. [Mar 29 06:27:43] DEBUG[11616][C-00000001] dahdi/bridge_native_dahdi.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: Cannot use native DAHDI. Channel 'Local/mobilephones@default-00000000;1' not compatible. [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Chose bridge technology simple_bridge [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7 is already using the new technology. [Mar 29 06:27:43] DEBUG[11616][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: 0x2114d98(PJSIP/161-00000000) is joining simple_bridge technology [Mar 29 06:27:43] DEBUG[11616][C-00000001] channel.c: Channel PJSIP/161-00000000 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11616][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;1 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11616][C-00000001] channel.c: Channel Local/mobilephones@default-00000000;1 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11616][C-00000001] channel.c: Channel PJSIP/161-00000000 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11616][C-00000001] channel.c: PJSIP/161-00000000: Topologies already match. Current: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> Requested: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:43] DEBUG[11616][C-00000001] channel.c: Local/mobilephones@default-00000000;1: Topologies already match. Current: <0:audio-0:audio:sendrecv (ulaw)> <1:video-1:video:sendrecv (h264)> Requested: <0:audio-0:audio:sendrecv (ulaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:43] DEBUG[20021] cdr.c: Finalized CDR for Local/mobilephones@default-00000000;1 - start 1648535253.721348 answer 1648535263.729274 end 1648535263.740315 dur 10.018 bill 0.011 dispo ANSWERED [Mar 29 06:27:43] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (369 bytes) from UDP:192.168.1.163:5060 ---> ACK sip:192.168.1.17:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK2001394111 From: "F-1-1-001" ;tag=1908330358 To: ;tag=435dfe33-def3-4ea2-bd8e-e8f5b062044f Call-ID: 911058807 CSeq: 21 ACK Contact: Max-Forwards: 70 User-Agent: DnakeVoip v1.0 Content-Length: 0 [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Terminated Inv State: CONNECTING [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Terminated Inv State: CONNECTING [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: Topology: Pending: (null topology) Active: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 Event: RX_MSG Inv State: CONFIRMED [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 Request: ACK [Mar 29 06:27:43] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 Handled request ACK ? yes [Mar 29 06:27:43] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x224f038) RTP 0x212c080 -- Received packet from 192.168.1.163:6200, dropping due to strict RTP protection. Qualifying new stream. [Mar 29 06:27:43] VERBOSE[11616][C-00000001] res_rtp_asterisk.c: 0x212c080 -- Strict RTP qualifying stream type: video [Mar 29 06:27:43] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x224f038) RTP 0x212c080 -- Received packet from 192.168.1.163:6200, dropping due to strict RTP protection. Will switch to it in 3 packets. [Mar 29 06:27:43] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x224f038) RTP 0x212c080 -- Received packet from 192.168.1.163:6200, dropping due to strict RTP protection. Will switch to it in 2 packets. [Mar 29 06:27:43] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x224f038) RTP 0x212c080 -- Received packet from 192.168.1.163:6200, dropping due to strict RTP protection. Will switch to it in 1 packets. [Mar 29 06:27:43] VERBOSE[11616][C-00000001] res_rtp_asterisk.c: 0x212c080 -- Strict RTP switching source address to 192.168.1.163:6200 [Mar 29 06:27:43] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x224f038) RTCP setting address on RTP instance [Mar 29 06:27:43] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x214a688) RTP ooh, format changed from none to h264 [Mar 29 06:27:43] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x214a688) RTCP starting transmission [Mar 29 06:27:43] VERBOSE[11683][C-00000001] bridge.c: Move-swap optimizing Local/mobilephones@default-00000000;2 <-- PJSIP/161-00000000. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Moving 0x2114d98(PJSIP/161-00000000) into bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1 swapping with Local/mobilephones@default-00000000;2 [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_channel.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: pulling 0x2114d98(PJSIP/161-00000000) [Mar 29 06:27:43] VERBOSE[11683][C-00000001] bridge_channel.c: Channel PJSIP/161-00000000 left 'simple_bridge' basic-bridge [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_channel.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: 0x2114d98(PJSIP/161-00000000) is leaving simple_bridge technology [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_channel.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: pushing 0x2114d98(PJSIP/161-00000000) by swapping with 0x2115108(Local/mobilephones@default-00000000;2) [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_channel.c: Setting 0x2115108(Local/mobilephones@default-00000000;2) state from:0 to:2 [Mar 29 06:27:43] DEBUG[20021] cdr.c: Finalized CDR for PJSIP/161-00000000 - start 1648535253.663076 answer 1648535263.731926 end 1648535263.881384 dur 10.218 bill 0.149 dispo ANSWERED [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_channel.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: pulling 0x2115108(Local/mobilephones@default-00000000;2) [Mar 29 06:27:43] VERBOSE[11683][C-00000001] bridge_channel.c: Channel Local/mobilephones@default-00000000;2 left 'simple_bridge' basic-bridge [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_channel.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: 0x2115108(Local/mobilephones@default-00000000;2) is leaving simple_bridge technology [Mar 29 06:27:43] VERBOSE[11683][C-00000001] bridge_channel.c: Channel PJSIP/161-00000000 swapped with Local/mobilephones@default-00000000;2 into 'simple_bridge' basic-bridge [Mar 29 06:27:43] DEBUG[20021] cdr.c: Finalized CDR for Local/mobilephones@default-00000000;2 - start 1648535253.722687 answer 1648535263.727692 end 1648535263.882154 dur 10.159 bill 0.154 dispo ANSWERED [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_native_rtp.c: Bridge 'c806adff-82ca-42f1-8093-5ff2be7e3cb1'. Checking compatability for channels 'PJSIP/141-00000001' and 'PJSIP/161-00000000' [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_native_rtp.c: Bridge 'c806adff-82ca-42f1-8093-5ff2be7e3cb1' can not use native RTP bridge as it was forbidden while getting details [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11683][C-00000001] dahdi/bridge_native_dahdi.c: Channel 'PJSIP/141-00000001' is not DAHDI. [Mar 29 06:27:43] DEBUG[11683][C-00000001] dahdi/bridge_native_dahdi.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: Cannot use native DAHDI. Channel 'PJSIP/141-00000001' not compatible. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Chose bridge technology simple_bridge [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1 is already using the new technology. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: 0x2114d98(PJSIP/161-00000000) is joining simple_bridge technology [Mar 29 06:27:43] DEBUG[11683][C-00000001] channel.c: Channel PJSIP/161-00000000 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11683][C-00000001] channel.c: Channel PJSIP/141-00000001 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11683][C-00000001] channel.c: Channel PJSIP/141-00000001 setting read format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11683][C-00000001] channel.c: Channel PJSIP/161-00000000 setting write format path: ulaw -> ulaw [Mar 29 06:27:43] DEBUG[11683][C-00000001] channel.c: PJSIP/161-00000000: Topologies already match. Current: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> Requested: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:43] DEBUG[11683][C-00000001] channel.c: PJSIP/141-00000001: Topologies already match. Current: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264)> Requested: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_native_rtp.c: Bridge 'fd2791bb-3e58-4978-9bc6-850dacedf3f7' can not use native RTP bridge as two channels are required [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11683][C-00000001] dahdi/bridge_native_dahdi.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: Cannot use native DAHDI. Must have two channels. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Chose bridge technology simple_bridge [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7 is already using the new technology. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_channel.c: Setting 0x2246c98(Local/mobilephones@default-00000000;1) state from:0 to:2 [Mar 29 06:27:43] DEBUG[11620][C-00000001] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Mar 29 06:27:43] DEBUG[11620][C-00000001] app_dial.c: Local/mobilephones@default-00000000;2: Done [Mar 29 06:27:43] DEBUG[11620][C-00000001] pbx.c: Spawn extension (default,mobilephones,3) exited non-zero on 'Local/mobilephones@default-00000000;2' [Mar 29 06:27:43] VERBOSE[11620][C-00000001] pbx.c: Spawn extension (default, mobilephones, 3) exited non-zero on 'Local/mobilephones@default-00000000;2' [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Soft-Hanging (0x10) up channel 'Local/mobilephones@default-00000000;2' [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Channel 0x2224cf8 'Local/mobilephones@default-00000000;2' hanging up. Refs: 2 [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel.c: Channel 0x2224cf8 'Local/mobilephones@default-00000000;2' destroying [Mar 29 06:27:43] DEBUG[11620][C-00000001] stasis.c: Destroying topic. name: channel:1648535253.2, detail: [Mar 29 06:27:43] DEBUG[11620][C-00000001] stasis.c: Topic 'channel:1648535253.2': 0x2132078 destroyed [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel_internal_api.c: : MultistreamFormats: (nothing) [Mar 29 06:27:43] DEBUG[11620][C-00000001] channel_internal_api.c: Channel is being initialized or destroyed [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_channel.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: pulling 0x2246c98(Local/mobilephones@default-00000000;1) [Mar 29 06:27:43] VERBOSE[11683][C-00000001] bridge_channel.c: Channel Local/mobilephones@default-00000000;1 left 'simple_bridge' basic-bridge [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge_channel.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: 0x2246c98(Local/mobilephones@default-00000000;1) is leaving simple_bridge technology [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: dissolving bridge with cause 16(Normal Clearing) [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: queueing action type:13 sub:1001 [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7 is dissolved, not performing smart bridge operation. [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: actually destroying basic bridge, nobody wants it anymore [Mar 29 06:27:43] DEBUG[11683][C-00000001] stasis.c: Destroying topic. name: bridge:all/bridge:fd2791bb-3e58-4978-9bc6-850dacedf3f7, detail: [Mar 29 06:27:43] DEBUG[11683][C-00000001] stasis.c: Topic 'bridge:all/bridge:fd2791bb-3e58-4978-9bc6-850dacedf3f7': 0x2255888 destroyed [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: calling basic bridge destructor [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: calling simple_bridge technology stop [Mar 29 06:27:43] DEBUG[11683][C-00000001] bridge.c: Bridge fd2791bb-3e58-4978-9bc6-850dacedf3f7: calling simple_bridge technology destructor [Mar 29 06:27:43] DEBUG[11683][C-00000001] channel.c: Channel 0x2221348 'Local/mobilephones@default-00000000;1' hanging up. Refs: 2 [Mar 29 06:27:43] DEBUG[11683][C-00000001] channel.c: Channel 0x2221348 'Local/mobilephones@default-00000000;1' destroying [Mar 29 06:27:43] DEBUG[20021] cdr.c: CDR for Local/mobilephones@default-00000000;1 is dialed and has no Party B; discarding [Mar 29 06:27:43] DEBUG[11683][C-00000001] stasis.c: Destroying topic. name: channel:1648535253.1, detail: [Mar 29 06:27:43] DEBUG[11683][C-00000001] stasis.c: Topic 'channel:1648535253.1': 0x221de98 destroyed [Mar 29 06:27:43] DEBUG[11683][C-00000001] channel_internal_api.c: : MultistreamFormats: (nothing) [Mar 29 06:27:43] DEBUG[11683][C-00000001] channel_internal_api.c: Channel is being initialized or destroyed [Mar 29 06:27:43] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x20e6e78) RTP 0x2250760 -- Received packet from 192.168.1.163:6000, dropping due to strict RTP protection. Qualifying new stream. [Mar 29 06:27:43] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP response (1100 bytes) from TLS:188.66.164.195:48138 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 5.37.239.43:5061;rport=5061;branch=z9hG4bKPj3a6d7111-10cc-4749-bfe0-56785b179871;alias From: "161" ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c To: ;tag=1994111823 Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 CSeq: 2242 INVITE Contact: Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.0.3.34 Session-Expires: 1800;refresher=uac Require: timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 449 v=0 o=141 8000 8000 IN IP4 10.169.147.148 s=SIP Call c=IN IP4 10.169.147.148 t=0 0 m=audio 26892 RTP/AVP 0 8 3 111 a=sendrecv a=rtcp:26893 IN IP4 10.169.147.148 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000[Mar 29 06:27:43] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP request (415 bytes) to TLS:188.66.164.195:48138 ---> ACK sip:141@188.66.164.195:48138;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 5.37.239.43:5061;rport;branch=z9hG4bKPj6ea03932-a53e-4923-b66f-33c8fd66ce34;alias From: "161" ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c To: ;tag=1994111823 Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 CSeq: 2242 ACK Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:43] VERBOSE[11616][C-00000001] res_rtp_asterisk.c: 0x2250760 -- Strict RTP qualifying stream type: audio [Mar 29 06:27:43] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x20e6e78) RTP 0x2250760 -- Received packet from 192.168.1.163:6000, dropping due to strict RTP protection. Will switch to it in 3 packets. [Mar 29 06:27:43] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x20e6e78) RTP 0x2250760 -- Received packet from 192.168.1.163:6000, dropping due to strict RTP protection. Will switch to it in 2 packets. [Mar 29 06:27:44] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x20e6e78) RTP 0x2250760 -- Received packet from 192.168.1.163:6000, dropping due to strict RTP protection. Will switch to it in 1 packets. [Mar 29 06:27:44] VERBOSE[11616][C-00000001] res_rtp_asterisk.c: 0x2250760 -- Strict RTP switching source address to 192.168.1.163:6000 [Mar 29 06:27:44] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x20e6e78) RTCP setting address on RTP instance [Mar 29 06:27:44] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP ooh, format changed from none to ulaw [Mar 29 06:27:44] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTCP starting transmission [Mar 29 06:27:44] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (800 bytes) from TLS:188.66.164.195:48138 ---> INFO sip:asterisk@5.37.239.43:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 10.169.147.148:45546;branch=z9hG4bK906554564;rport From: ;tag=1994111823 To: ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 CSeq: 2243 INFO Contact: Max-Forwards: 70 Supported: replaces, path, timer, 100rel, eventlist User-Agent: Grandstream Wave 1.0.3.34 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/media_control+xml Content-Length: 164 [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Request: INFO [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Handled request INFO ? yes [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Trying Inv State: CONFIRMED [Mar 29 06:27:44] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (347 bytes) to TLS:188.66.164.195:48138 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 10.169.147.148:45546;rport=48138;received=188.66.164.195;branch=z9hG4bK906554564 Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 From: ;tag=1994111823 To: ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c CSeq: 2243 INFO Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Completed Inv State: CONFIRMED [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Completed Inv State: CONFIRMED [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: Topology: Pending: (null topology) Active: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Completed Inv State: CONFIRMED [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: Topology: Pending: (null topology) Active: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:44] DEBUG[31646] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '192.168.1.163' [Mar 29 06:27:44] DEBUG[31646] res_pjsip/pjsip_resolver.c: Transport type for target '192.168.1.163' is 'UDP transport' [Mar 29 06:27:44] DEBUG[31646] res_pjsip/pjsip_resolver.c: Target '192.168.1.163' is an IP address, skipping resolution [Mar 29 06:27:44] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP request (597 bytes) to UDP:192.168.1.163:5060 ---> INFO sip:161@192.168.1.163:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPj949787a4-de33-4328-9f05-4fa5f66a683b From: ;tag=435dfe33-def3-4ea2-bd8e-e8f5b062044f To: "F-1-1-001" ;tag=1908330358 Call-ID: 911058807 CSeq: 17035 INFO Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Type: application/media_control+xml Content-Length: 178 [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Calling Inv State: CONFIRMED [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Calling Inv State: CONFIRMED [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: Topology: Pending: (null topology) Active: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:44] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP response (368 bytes) from UDP:192.168.1.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPj949787a4-de33-4328-9f05-4fa5f66a683b From: ;tag=435dfe33-def3-4ea2-bd8e-e8f5b062044f To: "F-1-1-001" ;tag=1908330358 Call-ID: 911058807 CSeq: 17035 INFO Contact: User-Agent: DnakeVoip v1.0 Content-Length: 0 [Mar 29 06:27:44] DEBUG[20027] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Terminated Inv State: CONFIRMED [Mar 29 06:27:44] DEBUG[20027] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:44] DEBUG[20027] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Terminated Inv State: CONFIRMED [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 Method: INFO Status: 200 [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:44] DEBUG[20027] res_pjsip_session.c: Topology: Pending: (null topology) Active: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Completed Inv State: CONFIRMED [Mar 29 06:27:44] DEBUG[20027] res_pjsip_session.c: [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000: INFO received final response code 200 [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Completed Inv State: CONFIRMED [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: Topology: Pending: (null topology) Active: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:44] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:44] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP 0x221e0c0 -- Received packet from 188.66.164.195:48139, dropping due to strict RTP protection. Qualifying new stream. [Mar 29 06:27:44] VERBOSE[11681][C-00000001] res_rtp_asterisk.c: 0x221e0c0 -- Strict RTP qualifying stream type: audio [Mar 29 06:27:44] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP 0x221e0c0 -- Received packet from 188.66.164.195:48139, dropping due to strict RTP protection. Will switch to it in 3 packets. [Mar 29 06:27:44] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP 0x221e0c0 -- Received packet from 188.66.164.195:48139, dropping due to strict RTP protection. Will switch to it in 2 packets. [Mar 29 06:27:44] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP 0x221e0c0 -- Received packet from 188.66.164.195:48139, dropping due to strict RTP protection. Will switch to it in 1 packets. [Mar 29 06:27:44] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (869 bytes) from TLS:212.146.160.239:53575 ---> REGISTER sip:Kindows.ddns.net:5061 SIP/2.0 Via: SIP/2.0/TLS 10.132.48.239:57390;branch=z9hG4bK747502227;rport;alias From: ;tag=24873490 To: Call-ID: 596385609-29210-1@BJC.BGI.B.CCJ CSeq: 2403 REGISTER Contact: ;reg-id=1;+sip.instance="" Authorization: Digest username="141", realm="asterisk", nonce="1648534772/7a2d52bbe19dfa690f1b46a02ea54009", uri="sip:Kindows.ddns.net:5061", response="4e6a60021c7d1a2c7fc995f36e9403be", algorithm=md5, cnonce="00446258", opaque="1ceba70333727d1d", qop=auth, nc=00000003 Max-Forwards: 70 User-Agent: Grandstream Wave 1.0.3.34 Supported: path Expires: 3600 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:44] DEBUG[31646] res_pjsip_nat.c: Saving contact '10.132.48.239:57390' [Mar 29 06:27:44] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (500 bytes) to TLS:212.146.160.239:53575 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 10.132.48.239:57390;rport=53575;received=212.146.160.239;branch=z9hG4bK747502227;alias Call-ID: 596385609-29210-1@BJC.BGI.B.CCJ From: ;tag=24873490 To: ;tag=z9hG4bK747502227 CSeq: 2403 REGISTER WWW-Authenticate: Digest realm="asterisk",nonce="1648535264/c31af14e66a67e559b559bc970111979",opaque="16e92f900c46c0ed",stale=true,algorithm=md5,qop="auth" Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:44] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP 0x221e0c0 -- Received packet from 188.66.164.195:48139, dropping due to strict RTP protection. Will switch to it in 3 packets. [Mar 29 06:27:44] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP 0x221e0c0 -- Received packet from 188.66.164.195:48139, dropping due to strict RTP protection. Will switch to it in 2 packets. [Mar 29 06:27:44] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (869 bytes) from TLS:212.146.160.239:53575 ---> REGISTER sip:Kindows.ddns.net:5061 SIP/2.0 Via: SIP/2.0/TLS 10.132.48.239:57390;branch=z9hG4bK547725859;rport;alias From: ;tag=24873490 To: Call-ID: 596385609-29210-1@BJC.BGI.B.CCJ CSeq: 2404 REGISTER Contact: ;reg-id=1;+sip.instance="" Authorization: Digest username="141", realm="asterisk", nonce="1648535264/c31af14e66a67e559b559bc970111979", uri="sip:Kindows.ddns.net:5061", response="38911c81bbea41a6f0938fd01036c784", algorithm=md5, cnonce="03665545", opaque="16e92f900c46c0ed", qop=auth, nc=00000001 Max-Forwards: 70 User-Agent: Grandstream Wave 1.0.3.34 Supported: path Expires: 3600 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:44] DEBUG[31646] res_pjsip_nat.c: Saving contact '10.132.48.239:57390' [Mar 29 06:27:44] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP 0x221e0c0 -- Received packet from 188.66.164.195:48139, dropping due to strict RTP protection. Will switch to it in 1 packets. [Mar 29 06:27:44] VERBOSE[31646] res_pjsip_registrar.c: Added contact 'sip:141@212.146.160.239:53575;transport=TLS;x-ast-orig-host=10.132.48.239:57390' to AOR '141' with expiration of 3600 seconds [Mar 29 06:27:44] DEBUG[11285] res_pjsip.c: 0x21919a8: Wrapper created [Mar 29 06:27:44] DEBUG[11285] res_pjsip.c: 0x21919a8: Set timer to 3000 msec [Mar 29 06:27:44] DEBUG[11285] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '212.146.160.239' [Mar 29 06:27:44] DEBUG[11285] res_pjsip/pjsip_resolver.c: Transport type for target '212.146.160.239' is 'TLS transport' [Mar 29 06:27:44] DEBUG[11285] res_pjsip/pjsip_resolver.c: Target '212.146.160.239' is an IP address, skipping resolution [Mar 29 06:27:44] VERBOSE[11285] res_pjsip_logger.c: <--- Transmitting SIP request (453 bytes) to TLS:212.146.160.239:53575 ---> OPTIONS sip:141@212.146.160.239:53575;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 5.37.239.43:5061;rport;branch=z9hG4bKPj9ebbb77f-a466-459a-ac05-582f4d7365f7;alias From: ;tag=b03a5637-f062-44f0-935c-fd568d456604 To: Contact: Call-ID: 4d043bfd-92be-482d-8e23-ea3e62e0112b CSeq: 26156 OPTIONS Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:44] DEBUG[31646] res_pjsip_nat.c: Restoring contact 188.66.164.195:48138 to 10.169.147.148:45546 [Mar 29 06:27:44] DEBUG[31646] res_pjsip_nat.c: Restoring contact 5.37.239.43:44166 to 192.168.1.190:44166 [Mar 29 06:27:44] DEBUG[31646] res_pjsip_nat.c: Restoring contact 212.146.160.239:53575 to 10.132.48.239:57390 [Mar 29 06:27:44] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (587 bytes) to TLS:212.146.160.239:53575 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 10.132.48.239:57390;rport=53575;received=212.146.160.239;branch=z9hG4bK547725859;alias Call-ID: 596385609-29210-1@BJC.BGI.B.CCJ From: ;tag=24873490 To: ;tag=z9hG4bK547725859 CSeq: 2404 REGISTER Date: Tue, 29 Mar 2022 06:27:44 GMT Contact: ;expires=3583 Contact: ;expires=3543 Contact: ;expires=3599 Expires: 3600 Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:44] DEBUG[11285] res_pjsip.c: 0x221b358: Wrapper created [Mar 29 06:27:44] DEBUG[11285] res_pjsip.c: 0x221b358: Set timer to 3000 msec [Mar 29 06:27:44] DEBUG[11285] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '5.37.239.43' [Mar 29 06:27:44] DEBUG[11285] res_pjsip/pjsip_resolver.c: Transport type for target '5.37.239.43' is 'TLS transport' [Mar 29 06:27:44] DEBUG[11285] res_pjsip/pjsip_resolver.c: Target '5.37.239.43' is an IP address, skipping resolution [Mar 29 06:27:44] VERBOSE[11285] res_pjsip_logger.c: <--- Transmitting SIP request (445 bytes) to TLS:5.37.239.43:44166 ---> OPTIONS sip:141@5.37.239.43:44166;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 5.37.239.43:5061;rport;branch=z9hG4bKPjd72d9ade-a3d4-4f12-a67d-c910fd1678c4;alias From: ;tag=d2496663-78d0-4fd3-8cce-2858d96761fa To: Contact: Call-ID: fa495654-350d-4ca2-be9f-da2f5737cde0 CSeq: 52856 OPTIONS Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:44] DEBUG[11285] res_pjsip.c: 0x2222718: Wrapper created [Mar 29 06:27:44] DEBUG[11285] res_pjsip.c: 0x2222718: Set timer to 3000 msec [Mar 29 06:27:44] DEBUG[11285] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '188.66.164.195' [Mar 29 06:27:44] DEBUG[11285] res_pjsip/pjsip_resolver.c: Transport type for target '188.66.164.195' is 'TLS transport' [Mar 29 06:27:44] DEBUG[11285] res_pjsip/pjsip_resolver.c: Target '188.66.164.195' is an IP address, skipping resolution [Mar 29 06:27:44] VERBOSE[11285] res_pjsip_logger.c: <--- Transmitting SIP request (451 bytes) to TLS:188.66.164.195:48138 ---> OPTIONS sip:141@188.66.164.195:48138;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 5.37.239.43:5061;rport;branch=z9hG4bKPjc742546c-38c4-4771-9428-d0292bed4859;alias From: ;tag=10f3b64d-9ae3-4ac9-bd94-5f67ea0683a9 To: Contact: Call-ID: 247e06f4-5567-4a6c-99d0-dd37a14427b0 CSeq: 56679 OPTIONS Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:44] VERBOSE[11681][C-00000001] res_rtp_asterisk.c: 0x221e0c0 -- Strict RTP switching source address to 188.66.164.195:48139 [Mar 29 06:27:44] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTCP setting address on RTP instance [Mar 29 06:27:44] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x20e6e78) RTP ooh, format changed from none to ulaw [Mar 29 06:27:44] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP response (494 bytes) from TLS:212.146.160.239:53575 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 5.37.239.43:5061;rport=5061;branch=z9hG4bKPj9ebbb77f-a466-459a-ac05-582f4d7365f7;alias From: ;tag=b03a5637-f062-44f0-935c-fd568d456604 To: ;tag=632641138 Call-ID: 4d043bfd-92be-482d-8e23-ea3e62e0112b CSeq: 26156 OPTIONS Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.0.3.34 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:45] DEBUG[31646] res_pjsip.c: 0x21919a8: PJSIP tsx response received [Mar 29 06:27:45] DEBUG[31646] res_pjsip.c: 0x21919a8: Callbacks executed [Mar 29 06:27:45] DEBUG[31646] res_pjsip.c: 0x21919a8: wrapper destroyed [Mar 29 06:27:45] VERBOSE[31646] res_pjsip/pjsip_options.c: Contact 141/sip:141@212.146.160.239:53575;transport=TLS;x-ast-orig-host=10.132.48.239:57390 is now Reachable. RTT: 50.733 msec [Mar 29 06:27:45] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP response (492 bytes) from TLS:188.66.164.195:48138 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 5.37.239.43:5061;rport=5061;branch=z9hG4bKPjc742546c-38c4-4771-9428-d0292bed4859;alias From: ;tag=10f3b64d-9ae3-4ac9-bd94-5f67ea0683a9 To: ;tag=47696835 Call-ID: 247e06f4-5567-4a6c-99d0-dd37a14427b0 CSeq: 56679 OPTIONS Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.0.3.34 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:45] DEBUG[31646] res_pjsip.c: 0x2222718: PJSIP tsx response received [Mar 29 06:27:45] DEBUG[31646] res_pjsip.c: 0x2222718: Callbacks executed [Mar 29 06:27:45] DEBUG[31646] res_pjsip.c: 0x2222718: wrapper destroyed [Mar 29 06:27:45] DEBUG[11681][C-00000001] stun.c: Scrambled STUN packet length (got 65356, expecting 0) [Mar 29 06:27:45] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP 0x221e0c0 -- Received packet from 188.66.164.195:48141, dropping due to strict RTP protection. Qualifying new stream. [Mar 29 06:27:45] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP 0x221e0c0 -- Received packet from 188.66.164.195:48141, dropping due to strict RTP protection. Will switch to it in 3 packets. [Mar 29 06:27:45] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP 0x221e0c0 -- Received packet from 188.66.164.195:48141, dropping due to strict RTP protection. Will switch to it in 2 packets. [Mar 29 06:27:45] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTP 0x221e0c0 -- Received packet from 188.66.164.195:48141, dropping due to strict RTP protection. Will switch to it in 1 packets. [Mar 29 06:27:45] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTCP got report of 20 bytes from 188.66.164.195:48142 [Mar 29 06:27:45] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTCP 0x221e0c0 -- from 188.66.164.195:48142: Failed first packet validity check [Mar 29 06:27:47] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x20e6e78) RTCP got report of 124 bytes from 192.168.1.163:6001 [Mar 29 06:27:47] DEBUG[20027] res_pjsip.c: 0x221b358: Internal tsx timer expired after 3000 msec [Mar 29 06:27:47] DEBUG[20027] res_pjsip.c: 0x221b358: Callbacks executed [Mar 29 06:27:48] DEBUG[11616][C-00000001] res_rtp_asterisk.c: (0x224f038) RTCP got report of 100 bytes from 192.168.1.163:6201 [Mar 29 06:27:48] VERBOSE[11681][C-00000001] res_rtp_asterisk.c: 0x221e0c0 -- Strict RTP learning complete - Locking on source address 188.66.164.195:48139 [Mar 29 06:27:48] VERBOSE[11616][C-00000001] res_rtp_asterisk.c: 0x212c080 -- Strict RTP learning complete - Locking on source address 192.168.1.163:6200 [Mar 29 06:27:48] VERBOSE[11616][C-00000001] res_rtp_asterisk.c: 0x2250760 -- Strict RTP learning complete - Locking on source address 192.168.1.163:6000 [Mar 29 06:27:49] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTCP got report of 52 bytes from 188.66.164.195:48144 [Mar 29 06:27:49] DEBUG[20027] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Terminated Inv State: CONFIRMED [Mar 29 06:27:49] DEBUG[20027] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:49] DEBUG[20027] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Terminated Inv State: CONFIRMED [Mar 29 06:27:49] DEBUG[20027] res_pjsip_session.c: Topology: Pending: (null topology) Active: <0:audio-0:audio:sendrecv (ulaw|alaw)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:49] DEBUG[20027] res_pjsip_session.c: [Mar 29 06:27:49] DEBUG[11681][C-00000001] res_rtp_asterisk.c: (0x2148e58) RTCP got report of 52 bytes from 188.66.164.195:48142 [Mar 29 06:27:51] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (585 bytes) from TLS:188.66.164.195:48138 ---> BYE sip:asterisk@5.37.239.43:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 10.169.147.148:45546;branch=z9hG4bK130787146;rport From: ;tag=1994111823 To: ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 CSeq: 2244 BYE Contact: Max-Forwards: 70 Supported: replaces, path, timer, 100rel, eventlist User-Agent: Grandstream Wave 1.0.3.34 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 [Mar 29 06:27:51] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (346 bytes) to TLS:188.66.164.195:48138 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 10.169.147.148:45546;rport=48138;received=188.66.164.195;branch=z9hG4bK130787146 Call-ID: a4138cf9-dff1-4552-9248-76ac9a14dc37 From: ;tag=1994111823 To: ;tag=95f5b64c-4c48-492a-908f-a2dbe266057c CSeq: 2244 BYE Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Completed Inv State: CONFIRMED [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Completed Inv State: CONFIRMED [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: Topology: Pending: (null topology) Active: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Event: TSX_STATE Inv State: DISCONNCTD [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Completed Inv State: DISCONNCTD [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 TSX State: Completed Inv State: DISCONNCTD [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: Topology: Pending: (null topology) Active: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g726)> <1:video-1:video:sendrecv (h264)> [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:51] DEBUG[31646] chan_pjsip.c: PJSIP/141-00000001 [Mar 29 06:27:51] DEBUG[31646] chan_pjsip.c: [Mar 29 06:27:51] DEBUG[11681][C-00000001] bridge_channel.c: Setting 0x224a198(PJSIP/141-00000001) state from:0 to:1 [Mar 29 06:27:51] DEBUG[11681][C-00000001] bridge_channel.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: pulling 0x224a198(PJSIP/141-00000001) [Mar 29 06:27:51] VERBOSE[11681][C-00000001] bridge_channel.c: Channel PJSIP/141-00000001 left 'simple_bridge' basic-bridge [Mar 29 06:27:51] DEBUG[11681][C-00000001] bridge_channel.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: 0x224a198(PJSIP/141-00000001) is leaving simple_bridge technology [Mar 29 06:27:51] DEBUG[11681][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: dissolving bridge with cause 16(Normal Clearing) [Mar 29 06:27:51] DEBUG[11681][C-00000001] bridge_channel.c: Setting 0x2114d98(PJSIP/161-00000000) state from:0 to:2 [Mar 29 06:27:51] DEBUG[11681][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: queueing action type:13 sub:1001 [Mar 29 06:27:51] DEBUG[11681][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1 is dissolved, not performing smart bridge operation. [Mar 29 06:27:51] DEBUG[11681][C-00000001] channel.c: Channel 0x2134a98 'PJSIP/141-00000001' hanging up. Refs: 2 [Mar 29 06:27:51] DEBUG[20021] cdr.c: Finalized CDR for PJSIP/161-00000000 - start 1648535263.882948 answer 1648535263.882948 end 1648535271.357414 dur 7.474 bill 7.474 dispo ANSWERED [Mar 29 06:27:51] DEBUG[11681][C-00000001] chan_pjsip.c: PJSIP/141-00000001 [Mar 29 06:27:51] DEBUG[11681][C-00000001] chan_pjsip.c: AST hangup cause 16 (no match found in PJSIP) [Mar 29 06:27:51] DEBUG[11681][C-00000001] chan_pjsip.c: Cause: 0 [Mar 29 06:27:51] DEBUG[31646] chan_pjsip.c: PJSIP/141-00000001 [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: PJSIP/141-00000001 Response 0 [Mar 29 06:27:51] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE RTP transport deallocating [Mar 29 06:27:51] DEBUG[31646] res_rtp_asterisk.c: (0x2148e58) ICE stopped [Mar 29 06:27:51] DEBUG[11616][C-00000001] bridge_channel.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: pulling 0x2114d98(PJSIP/161-00000000) [Mar 29 06:27:51] VERBOSE[11616][C-00000001] bridge_channel.c: Channel PJSIP/161-00000000 left 'simple_bridge' basic-bridge [Mar 29 06:27:51] DEBUG[31646] rtp_engine.c: Destroyed RTP instance '0x2148e58' [Mar 29 06:27:51] DEBUG[11616][C-00000001] bridge_channel.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: 0x2114d98(PJSIP/161-00000000) is leaving simple_bridge technology [Mar 29 06:27:51] DEBUG[11616][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1 is dissolved, not performing smart bridge operation. [Mar 29 06:27:51] DEBUG[11616][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: actually destroying basic bridge, nobody wants it anymore [Mar 29 06:27:51] DEBUG[11616][C-00000001] stasis.c: Destroying topic. name: bridge:all/bridge:c806adff-82ca-42f1-8093-5ff2be7e3cb1, detail: [Mar 29 06:27:51] DEBUG[11616][C-00000001] stasis.c: Topic 'bridge:all/bridge:c806adff-82ca-42f1-8093-5ff2be7e3cb1': 0x2226108 destroyed [Mar 29 06:27:51] DEBUG[11616][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: calling basic bridge destructor [Mar 29 06:27:51] DEBUG[11616][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: calling simple_bridge technology stop [Mar 29 06:27:51] DEBUG[11616][C-00000001] bridge.c: Bridge c806adff-82ca-42f1-8093-5ff2be7e3cb1: calling simple_bridge technology destructor [Mar 29 06:27:51] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) ICE RTP transport deallocating [Mar 29 06:27:51] DEBUG[11616][C-00000001] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Mar 29 06:27:51] DEBUG[11616][C-00000001] app_dial.c: PJSIP/161-00000000: Done [Mar 29 06:27:51] DEBUG[11616][C-00000001] pbx.c: Spawn extension (fullrights,601,3) exited non-zero on 'PJSIP/161-00000000' [Mar 29 06:27:51] VERBOSE[11616][C-00000001] pbx.c: Spawn extension (fullrights, 601, 3) exited non-zero on 'PJSIP/161-00000000' [Mar 29 06:27:51] DEBUG[31646] res_rtp_asterisk.c: (0x214a688) ICE stopped [Mar 29 06:27:51] DEBUG[11616][C-00000001] channel.c: Soft-Hanging (0x10) up channel 'PJSIP/161-00000000' [Mar 29 06:27:51] DEBUG[11616][C-00000001] channel.c: Channel 0x2257be8 'PJSIP/161-00000000' hanging up. Refs: 2 [Mar 29 06:27:51] DEBUG[11616][C-00000001] chan_pjsip.c: PJSIP/161-00000000 [Mar 29 06:27:51] DEBUG[11616][C-00000001] chan_pjsip.c: AST hangup cause 16 (no match found in PJSIP) [Mar 29 06:27:51] DEBUG[31646] rtp_engine.c: Destroyed RTP instance '0x214a688' [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:51] DEBUG[31646] channel.c: Channel 0x2134a98 'PJSIP/141-00000001' destroying [Mar 29 06:27:51] DEBUG[20021] cdr.c: CDR for PJSIP/141-00000001 is dialed and has no Party B; discarding [Mar 29 06:27:51] DEBUG[31646] stasis.c: Destroying topic. name: channel:1648535259.3, detail: [Mar 29 06:27:51] DEBUG[31646] stasis.c: Topic 'channel:1648535259.3': 0x2134688 destroyed [Mar 29 06:27:51] DEBUG[31646] channel_internal_api.c: : MultistreamFormats: (nothing) [Mar 29 06:27:51] DEBUG[31646] channel_internal_api.c: Channel is being initialized or destroyed [Mar 29 06:27:51] DEBUG[20027] res_pjsip_session.c: 141 TSX State: Terminated Inv State: DISCONNCTD [Mar 29 06:27:51] DEBUG[31646] chan_pjsip.c: [Mar 29 06:27:51] DEBUG[11616][C-00000001] chan_pjsip.c: Cause: 0 [Mar 29 06:27:51] DEBUG[11284] chan_pjsip.c: PJSIP/161-00000000 [Mar 29 06:27:51] DEBUG[11284] res_pjsip_session.c: PJSIP/161-00000000 Response 0 [Mar 29 06:27:51] DEBUG[11284] res_rtp_asterisk.c: (0x20e6e78) ICE RTP transport deallocating [Mar 29 06:27:51] DEBUG[11284] rtp_engine.c: Destroyed RTP instance '0x20e6e78' [Mar 29 06:27:51] DEBUG[20027] res_pjsip_session.c: Disconnected [Mar 29 06:27:51] DEBUG[11284] res_rtp_asterisk.c: (0x224f038) ICE RTP transport deallocating [Mar 29 06:27:51] DEBUG[20027] res_pjsip_session.c: (null session) TSX State: Terminated Inv State: DISCONNCTD [Mar 29 06:27:51] DEBUG[20027] res_pjsip_session.c: [Mar 29 06:27:51] DEBUG[11284] rtp_engine.c: Destroyed RTP instance '0x224f038' [Mar 29 06:27:51] DEBUG[11284] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '192.168.1.163' [Mar 29 06:27:51] DEBUG[11284] res_pjsip/pjsip_resolver.c: Transport type for target '192.168.1.163' is 'UDP transport' [Mar 29 06:27:51] DEBUG[11284] res_pjsip/pjsip_resolver.c: Target '192.168.1.163' is an IP address, skipping resolution [Mar 29 06:27:51] VERBOSE[11284] res_pjsip_logger.c: <--- Transmitting SIP request (393 bytes) to UDP:192.168.1.163:5060 ---> BYE sip:161@192.168.1.163:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPjf9f239bf-d6d0-4479-8f1a-14829114a0a3 From: ;tag=435dfe33-def3-4ea2-bd8e-e8f5b062044f To: "F-1-1-001" ;tag=1908330358 Call-ID: 911058807 CSeq: 17036 BYE Reason: Q.850;cause=16 Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:51] DEBUG[11284] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Calling Inv State: CONFIRMED [Mar 29 06:27:51] DEBUG[11284] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:51] DEBUG[11284] res_pjsip_session.c: PJSIP/161-00000000 TSX State: Calling Inv State: CONFIRMED [Mar 29 06:27:51] DEBUG[11284] res_pjsip_session.c: Topology: Pending: (null topology) Active: (null topology) [Mar 29 06:27:51] DEBUG[11284] res_pjsip_session.c: [Mar 29 06:27:51] DEBUG[11284] res_pjsip_session.c: [Mar 29 06:27:51] DEBUG[11284] channel.c: Channel 0x2257be8 'PJSIP/161-00000000' destroying [Mar 29 06:27:51] DEBUG[11284] stasis.c: Destroying topic. name: channel:1648535253.0, detail: [Mar 29 06:27:51] DEBUG[11284] stasis.c: Topic 'channel:1648535253.0': 0x2254ca8 destroyed [Mar 29 06:27:51] DEBUG[11284] channel_internal_api.c: : MultistreamFormats: (nothing) [Mar 29 06:27:51] DEBUG[11284] channel_internal_api.c: Channel is being initialized or destroyed [Mar 29 06:27:51] DEBUG[11284] chan_pjsip.c: [Mar 29 06:27:51] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP response (328 bytes) from UDP:192.168.1.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPjf9f239bf-d6d0-4479-8f1a-14829114a0a3 From: ;tag=435dfe33-def3-4ea2-bd8e-e8f5b062044f To: "F-1-1-001" ;tag=1908330358 Call-ID: 911058807 CSeq: 17036 BYE User-Agent: DnakeVoip v1.0 Content-Length: 0 [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: 161 Method: BYE Status: 200 [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: 161 Event: TSX_STATE Inv State: DISCONNCTD [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: 161 TSX State: Completed Inv State: DISCONNCTD [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: 161: BYE received final response code 200 [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: Nothing delayed [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: 161 TSX State: Completed Inv State: DISCONNCTD [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: Topology: Pending: (null topology) Active: (null topology) [Mar 29 06:27:51] DEBUG[31646] res_pjsip_session.c: [Mar 29 06:27:51] DEBUG[31646] chan_pjsip.c: 161 [Mar 29 06:27:51] DEBUG[31646] chan_pjsip.c: No channel [Mar 29 06:27:55] VERBOSE[11548] asterisk.c: Remote UNIX connection disconnected [Mar 29 06:27:56] VERBOSE[11285] res_pjsip/pjsip_options.c: Contact 141/sip:141@212.146.160.239:53575;transport=TLS;x-ast-orig-host=10.132.48.239:57390 has been deleted [Mar 29 06:27:56] VERBOSE[31646] res_pjsip_registrar.c: Removed contact 'sip:141@212.146.160.239:53575;transport=TLS;x-ast-orig-host=10.132.48.239:57390' from AOR '141' due to shutdown [Mar 29 06:27:56] DEBUG[20027] res_pjsip_session.c: 161 TSX State: Terminated Inv State: DISCONNCTD [Mar 29 06:27:56] DEBUG[20027] res_pjsip_session.c: Disconnected [Mar 29 06:27:56] DEBUG[20027] res_pjsip_session.c: (null session) TSX State: Terminated Inv State: DISCONNCTD [Mar 29 06:27:56] DEBUG[20027] res_pjsip_session.c: [Mar 29 06:27:57] DEBUG[31646] res_pjsip.c: 0x20fd9e8: Wrapper created [Mar 29 06:27:57] DEBUG[31646] res_pjsip.c: 0x20fd9e8: Set timer to 3000 msec [Mar 29 06:27:57] DEBUG[31646] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '192.168.1.163' [Mar 29 06:27:57] DEBUG[31646] res_pjsip/pjsip_resolver.c: Transport type for target '192.168.1.163' is 'UDP transport' [Mar 29 06:27:57] DEBUG[31646] res_pjsip/pjsip_resolver.c: Target '192.168.1.163' is an IP address, skipping resolution [Mar 29 06:27:57] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP request (458 bytes) to UDP:192.168.1.163:5060 ---> OPTIONS sip:161@192.168.1.163:5060;line=e63b8695feffcc4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.17:5060;rport;branch=z9hG4bKPj5fc32f58-9375-4847-936d-155ad526acfe From: ;tag=ef248ab7-7174-494e-b5f6-4449649f5c7c To: Contact: Call-ID: 328b7dc7-c44f-4c77-9d21-3fe17feb9c9f CSeq: 43259 OPTIONS Max-Forwards: 70 User-Agent: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:27:57] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP response (368 bytes) from UDP:192.168.1.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.17:5060;rport=5060;branch=z9hG4bKPj5fc32f58-9375-4847-936d-155ad526acfe From: ;tag=ef248ab7-7174-494e-b5f6-4449649f5c7c To: ;tag=890977271 Call-ID: 328b7dc7-c44f-4c77-9d21-3fe17feb9c9f CSeq: 43259 OPTIONS User-Agent: DnakeVoip v1.0 Content-Length: 0 [Mar 29 06:27:57] DEBUG[31646] res_pjsip.c: 0x20fd9e8: PJSIP tsx response received [Mar 29 06:27:57] DEBUG[31646] res_pjsip.c: 0x20fd9e8: Callbacks executed [Mar 29 06:27:57] DEBUG[31646] res_pjsip.c: 0x20fd9e8: wrapper destroyed [Mar 29 06:28:00] DEBUG[20027] res_pjsip.c: 0x200b108: PJSIP tsx timer expired [Mar 29 06:28:00] DEBUG[20027] res_pjsip.c: 0x200b108: wrapper destroyed [Mar 29 06:28:06] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (301 bytes) from UDP:192.168.1.163:5060 ---> OPTIONS sip:192.168.1.17 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1327472284 From: ;tag=1635917703 To: Call-ID: 509392734 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 70 User-Agent: DnakeVoip v1.0 Content-Length: 0 [Mar 29 06:28:06] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (446 bytes) to UDP:192.168.1.163:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1327472284 Call-ID: 509392734 From: ;tag=1635917703 To: ;tag=z9hG4bK1327472284 CSeq: 20 OPTIONS WWW-Authenticate: Digest realm="asterisk",nonce="1648535286/6eb9d3a1ff40e63e65712c1f91160312",opaque="14efdefe1c44457f",algorithm=md5,qop="auth" Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:28:06] VERBOSE[20027] res_pjsip_logger.c: <--- Received SIP request (563 bytes) from UDP:192.168.1.163:5060 ---> OPTIONS sip:192.168.1.17 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.163:5060;rport;branch=z9hG4bK1040850458 From: ;tag=1635917703 To: Call-ID: 509392734 CSeq: 21 OPTIONS Authorization: Digest username="161", realm="asterisk", nonce="1648535286/6eb9d3a1ff40e63e65712c1f91160312", uri="sip:192.168.1.17", response="b6e94903f6465d5b97fe55ed38ea008a", algorithm=MD5, cnonce="0a4f113b", opaque="14efdefe1c44457f", qop=auth, nc=00000001 Accept: application/sdp Max-Forwards: 70 User-Agent: DnakeVoip v1.0 Content-Length: 0 [Mar 29 06:28:06] VERBOSE[31646] res_pjsip_logger.c: <--- Transmitting SIP response (775 bytes) to UDP:192.168.1.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.163:5060;rport=5060;received=192.168.1.163;branch=z9hG4bK1040850458 Call-ID: 509392734 From: ;tag=1635917703 To: ;tag=z9hG4bK1040850458 CSeq: 21 OPTIONS Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Accept-Encoding: identity Accept-Language: en Server: SHAULA-001(7.4.0) Content-Length: 0 [Mar 29 06:28:11] DEBUG[11682] threadpool.c: Worker thread idle timeout reached. Dying. [Mar 29 06:28:11] DEBUG[11615] threadpool.c: Worker thread idle timeout reached. Dying.