Below are some sample configurations to demonstrate various scenarios with complete pjsip.conf files. To see examples side by side with old chan_sip config head to Migrating from chan_sip to res_pjsip. Explanations of the config sections found in each example can be found in PJSIP Configuration Sections and Relationships.
A tutorial on secure and encrypted calling is located in the Secure Calling section of the wiki.
An endpoint with a single SIP phone with inbound registration to Asterisk
A SIP trunk to your service provider, including outbound registration
Multiple endpoints with phones registering to Asterisk, using templates
Comments:
Are names like [auth6001] mandatory or I can simply use just [6001] as with aor, endpoint etc? Edit: Seems like the latter is the case. ![]() |
That is answered here: PJSIP Configuration Sections and Relationships#SectionNames Thanks! ![]() |