git.asterisk.org Git - asterisk/asterisk.git/history - configs/samples ... Learn more 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. To change your SIP port to 5160: Do one of the following: Go to /etc/asterisk/, or. No voice transmission, PJSIP behind NAT - Stack Overflow 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. ; First, manually written examples to serve as a handy reference. Q&A for work. The trunk seems to always negotiate to G729, so Asterisk ends up transcoding the ulaw to G729 between the two, and faxes have lots of issues. Disable internal connection oriented keep-alive." into 15. commit | commitdiff | tree. In the PBX web interface, edit the Trunk Peer Details in your system's web interface by . res_pjsip: Change log message from error to warning for valid use cases The PJSIP Configuration Wizard ⋆ Asterisk Sorcery. 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 Example of old configuration: alwaysauthreject=yes domain = asterisk.mydomain.com allowexternaldomains = no allowguest = no deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/255.0.0.0 And as the author of the question wrote, you have to use a firewall and fail2ban. Only 5160 (which is for chan_sip) I had this working in chan_sip and using TIPCon1 soft-phone ( TIPcon1 download | SourceForge.net ). This has meant that to enable PJSIP support in Asterisk you have needed to install PJPROJECT yourself using some method. A recent change attempted to optimize startup by not updating contact status. PJSIP-pjproject - Asterisk Project - Asterisk Project Wiki If you really don't want to wait for the "time out" then you should be able to delete them using the following: *CLI> database deltree registrar/contact. So either the endpoint (device) needs to explicitly unregister, or you can wait for the registration to "time out". In the asterisk advanced settings there is a section for device defaults. 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. ael set debug {read|tokens|macros|contexts|off} -- Enable AEL debugging flags. The channel driver itself being chan_pjsip which depends on res_pjsip and its many associated modules. FreePBX disabling modules for pjsip - FreePBX Community Forums it is adding the following lines: noload = chan_pjsip.so noload = res_pjsip_endpoint_identifier_anonymous.so noload = res_pjsip_messaging.so noload = res_pjsip_pidf.so noload = res_pjsip_session.so noload = func_pjsip_endpoint.so . Quarea (Quarea) May 21, 2019, 3:55pm #20 This option only applies to chan_sip devices. res_sorcery_astdb: Filter fields to only the registered ones.