search for: pjsip_configur

...ency=60 default_expiration=120 maximum_expiration=600 minimum_expiration=90 log: [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added contact 'sip:17367 at 46.39.229.18:37910' to AOR '17367' with expiration of 90 seconds [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:37910 has been created [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:27143 has been deleted [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:1736...

Hello! Today, I tried to switch from asterisk 13.7.2 to 13.9, but I'm getting strange problem w/ the registering of all of my extensions. It looks like that: [2016-05-12 08:59:38] VERBOSE[2332] res_pjsip/pjsip_configuration.c: Contact 107/sip:107 at 192.168.15.73:5060 has been created [2016-05-12 08:59:38] VERBOSE[2272] res_pjsip_registrar.c: Added contact 'sip:107 at 192.168.15.73:5060' to AOR '107' with expiration of 180 seconds [2016-05-12 08:59:38] VERBOSE[2332] res_pjsip/pjsip_configuration...

ping times are fine as well: [root at freepbx asterisk]# ping sipgate.de PING sipgate.de (217.10.79.9) 56(84) bytes of data. 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57

Hello everyone. I allow myself to submit a problem that I can not solve with my VOIP provider Orange in France [2023-06-08 13:19:03] ERROR[185091]: res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid character '@' [2023-06-08 13:19:03] ERROR[185091]: config_options.c:798 aco_process_var: Error parsing from_user=75B55BTQUHSG at orange-obs.fr at line 0 of   == c...

...sip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. - configs: Fix typo in pjsip.conf.sample. - res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown - res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters - .github: PRSubmitActions: Fix adding reviewers...

...sip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. - configs: Fix typo in pjsip.conf.sample. - res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown - res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters - .github: PRSubmitActions: Fix adding reviewers...

...sip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. - configs: Fix typo in pjsip.conf.sample. - res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown - res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters - .github: PRSubmitActions: Fix adding reviewers...

...sip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. - configs: Fix typo in pjsip.conf.sample. - res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown - res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters - .github: PRSubmitActions: Fix adding reviewers...

...sip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. - configs: Fix typo in pjsip.conf.sample. - res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown - res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters - .github: PRSubmitActions: Fix adding reviewers...

...sip: Include cipher limit in config error message. - res_speech: allow speech to translate input channel - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation. - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header. - api.wiki.mustache: Fix indentation in generated markdown - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. - configs: Fix typo in pjsip.conf.sample. - res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown - res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters - .github: PRSubmitActions: Fix adding reviewers...