res_pjsip: Change log message from error to warning for valid use cases If you're not already using chan_pjsip, now is a great time to begin trying, testing, and deploying chan_pjsip. Migrating from chan_sip to res_pjsip - Asterisk Project Wiki ael set debug {read|tokens|macros|contexts|off} -- Enable AEL debugging flags. To change your SIP port to 5160: Do one of the following: Go to /etc/asterisk/, or. Hi, I am using both sip and pjsip extensions on my Asterisk setup. 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. git.asterisk.org Git - asterisk/asterisk.git/blob - res/res_pjsip.c We also recommend checking which version of Asterisk your PBX is based on, as there are significant differences between each revision. The first day, I made my configurations and all chan_sip and chan_pjsip extensions were working fine. asterisk console commands. In the PBX web interface, edit the Trunk Peer Details in your system's web interface by . git.asterisk.org Git - asterisk/asterisk.git/blob - configs/samples ... next page → res_sorcery_astdb: Filter fields to only the registered ones. -- Execute a shell command. Set stun_ignore_failure to PJ_FALSE (by directly modifying the code as there is no param to disable it). I'm trying write softphone app with pjsua. Asterisk 14: Coming with improved PJSIP DNS Support! Asterisk 13.8.0: Now With Easier PJSIP Install Method! res_pjsip: Default endpoints to the "offline" status. Bundling allows a self-contained PJSIP to exist within Asterisk and be used by all functionality within it.