The named pickup groups that a channel can pickup. 'f.example.com' and 'foo..com' are not allowed. I'm using chan_pjsip trunks so I'll try to find where to add the "session-timers=refuse" in the trunk configuration, or I'll change the trunk to chan_sip. It's safer to just restart Asterisk clean. Since this essentially replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be specified in the endpoint's allowed codec list. SIP provider will call your server with a user name of "mytrunk". RFC 3261 says that the response to an OPTIONS request MUST be the same had the request been an INVITE. Setting the value to zero disables the timeout. jcolp November 21, 2021, 2:37pm #2 PJSIP doesn't have an automatic transport. Whitespace is ignored and they may be specified in any order. This effectively makes the semicolon a non-usable character for PJSIP endpoint names, extensions, and AORs. This option applies when an external entity subscribes to an AoR for Message Waiting Indications. Endpoints and AORs can be identified in multiple ways. Determines whether media may flow directly between endpoints. By default this option is set to 0, which means do not check. Determines whether res_pjsip will use and enforce usage of AVP, regardless of the RTP profile in use for this endpoint. This will force the endpoint to use the specified transport configuration to send SIP messages. Respond to a SIP invite with the single most preferred codec rather than advertising all joint codec capabilities. Initial number of threads in the res_pjsip threadpool. FreePBX Disabling PJSIP and Changing SIP Default port - YouTube Set transaction timer B value (milliseconds). This option only applies if media_encryption is set to sdes or dtls. This may result in a delay before an attack is recognized. PJSIP is the new channel library for Asterisk, replacing the older DAHDI and LIBPRI drivers. If set to no, res_pjsip will use the AVP or SAVP RTP profile for all media offers on outbound calls and media updates, and will decline media offers not using the AVP or SAVP profile. The caller-id and redirecting number strings obtained from incoming SIP URI user fields are always truncated at the first semicolon. The minimum allowed expiry time for subscriptions initiated by the endpoint.