As of Asterisk 13.8.0 another simpler option will be available instead: bundling. FreePBX disabling modules for pjsip - FreePBX Community Forums atl*CLI> core show help. git.asterisk.org Git - asterisk/asterisk.git/blob - configs/samples ... app_voicemail mailboxes must be specified as mailbox@context; for example: mailboxes=6001@default. Once pjsua is up, shutdown (connection . That worked for me at least. uri_pjsip mailboxes Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state changes happen for any of the specified mailboxes. If Asterisk is already running you can unload chan_sip using "module unload chan_sip.so" from the console, but if it started before PJSIP then it would cause problems. Beginning with Asterisk 15.0.0, it is enabled by default but can be disabled with the --without-pjproject-bundled option to ./configure. Q&A for work. ; reference to jog your memory when you need to write up a new configuration. res_pjsip: Change log message from error to warning for valid use cases ael set debug {read|tokens|macros|contexts|off} -- Enable AEL debugging flags. ; This file has two main sections. The output should look like the following: Module Description Use Count Status Support Level res_pjsip_endpoint_identifier_anonymous.so PJSIP Anonymous endpoint identifier 0 Running core Ensure that the "anonymous" endpoint has been properly loaded. Learn more projects / asterisk/asterisk.git / history commit grep author committer pickaxe ? Asterisk new PJSIP driver security option - Server Fault ! The instructions below are meant to assist you with the basic configuration of Asterisk (PJSIP). CHAN_SIP / CHAN_PJSIP 401 Unauthorized on INVITE ncinta December 20, 2021, 7:26pm #1 Brief Description - Asterisk supports RTT (real time text) with an addition of a couple statements in the configuration files, at least using chan_sip. Teams. CHAN_SIP / CHAN_PJSIP 401 Unauthorized on INVITE - Asterisk SIP ... git.asterisk.org Git - asterisk/asterisk.git/log It was done in a generic fashion though so other modules could use it and additional . When PJSIP was being written it was decided that a new data (not specifically configuration) layer would be written. Clearing Unavailable PJSIP Registrations - Asterisk SIP - Asterisk ... Quarea (Quarea) May 21, 2019, 3:55pm #20 This option only applies to chan_sip devices. The code even accounts for contacts/AORs But sometimes FreePBX is disabling my pjsip modules at startup by modifying the modules.conf. PSA: chan_sip status changed to "deprecated" & Asterisk 17.0.0-rc2 Release I had this working in chan_sip and using TIPCon1 soft-phone ( TIPcon1 download | SourceForge.net ). PJSIP Configuration Design ⋆ Asterisk lordaker March 15, 2018, 2:50pm #5. since I'm not able to organically reproduce the bug, to test it you can disable pjsip by hand: From FreePBX interface, open "Settings" > "Advanced Settings" find "SIP Channel Driver" variable and set it to "chan_sip" Submit and apply changes Now you should be able to verify the bug condition with grep pjsip /etc/asterisk/modules.conf
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