diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample index 4c42e8a5fa..99bdfb99d0 100644 --- a/configs/samples/pjsip.conf.sample +++ b/configs/samples/pjsip.conf.sample @@ -672,7 +672,7 @@ ; usage of media encryption for this endpoint (default: ; "no") ;media_encryption_optimistic=no ; Use encryption if possible but don't fail the call - ; if not possible. + ; if not possible. ;g726_non_standard=no ; When set to "yes" and an endpoint negotiates g.726 ; audio then g.726 for AAL2 packing order is used contrary ; to what is recommended in RFC3551. Note, 'g726aal2' also @@ -752,7 +752,7 @@ ;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80 ; byte tags (default: "no") ;set_var= ; Variable set on a channel involving the endpoint. For multiple - ; channel variables specify multiple 'set_var'(s) + ; channel variables specify multiple 'set_var'(s) ;rtp_keepalive= ; Interval, in seconds, between comfort noise RTP packets if ; RTP is not flowing. This setting is useful for ensuring that ; holes in NATs and firewalls are kept open throughout a call. @@ -794,7 +794,7 @@ ; (default: "") ;ca_list_path= ; Path to directory containing certificates to read TLS ONLY. ; PJProject version 2.4 or higher is required for this option to - ; be used. + ; be used. ; (default: "") ;cert_file= ; Certificate file for endpoint TLS ONLY ; Will read .crt or .pem file but only uses cert, @@ -886,8 +886,8 @@ ;disable_tcp_switch=yes ; Disable automatic switching from UDP to TCP transports ; if outgoing request is too large. ; See RFC 3261 section 18.1.1. - ; Disabling this option has been known to cause interoperability - ; issues, so disable at your own risk. + ; Disabling this option has been known to cause interoperability + ; issues, so disable at your own risk. ; (default: "yes") ;type= ; Must be of type system (default: "") @@ -917,10 +917,10 @@ ;contact_expiration_check_interval=30 ; The interval (in seconds) to check for expired contacts. ;disable_multi_domain=no - ; Disable Multi Domain support. - ; If disabled it can improve realtime performace by reducing - ; number of database requsts - ; (default: "no") + ; Disable Multi Domain support. + ; If disabled it can improve realtime performace by reducing + ; number of database requsts + ; (default: "no") ;endpoint_identifier_order=ip,username,anonymous ; The order by which endpoint identifiers are given priority. ; Currently, "ip", "username", "auth_username" and "anonymous" are valid diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h index 9dd70dbcaa..9bb2a82c5d 100644 --- a/include/asterisk/res_pjsip.h +++ b/include/asterisk/res_pjsip.h @@ -749,9 +749,9 @@ struct ast_sip_endpoint { unsigned int usereqphone; /*! Whether to pass through hold and unhold using re-invites with recvonly and sendrecv */ unsigned int moh_passthrough; - /* Access control list */ + /*! Access control list */ struct ast_acl_list *acl; - /* Restrict what IPs are allowed in the Contact header (for registration) */ + /*! Restrict what IPs are allowed in the Contact header (for registration) */ struct ast_acl_list *contact_acl; /*! The number of seconds into call to disable fax detection. (0 = disabled) */ unsigned int faxdetect_timeout; diff --git a/res/res_pjsip.c b/res/res_pjsip.c index a1deb9e5fc..f56790df70 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -217,10 +217,9 @@ DTMF is sent as SIP INFO packets. - - DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not. - - + + DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not. + @@ -510,15 +509,15 @@ Force g.726 to use AAL2 packing order when negotiating g.726 audio - When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2 - packing order instead of what is recommended by RFC3551. Since this essentially - replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be - specified in the endpoint's allowed codec list. + When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2 + packing order instead of what is recommended by RFC3551. Since this essentially + replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be + specified in the endpoint's allowed codec list. Determines whether chan_pjsip will indicate ringing using inband - progress. + progress. If set to yes, chan_pjsip will send a 183 Session Progress when told to indicate ringing and will immediately start sending ringing @@ -814,7 +813,7 @@ Variable set on a channel involving the endpoint. - When a new channel is created using the endpoint set the specified + When a new channel is created using the endpoint set the specified variable(s) on that channel. For multiple channel variables specify multiple 'set_var'(s). @@ -1455,9 +1454,9 @@ Value used in User-Agent header for SIP requests and Server header for SIP responses. - When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given - peer who registers or unregisters with us. - + When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given + peer who registers or unregisters with us. + Endpoint to use when sending an outbound request to a URI without a specified endpoint. @@ -1466,15 +1465,15 @@ Enable/Disable SIP debug logging. Valid options include yes|no or - a host address + a host address The order by which endpoint identifiers are processed and checked. - Identifier names are usually derived from and can be found in the endpoint - identifier module itself (res_pjsip_endpoint_identifier_*). - You can use the CLI command "pjsip show identifiers" to see the - identifiers currently available. - + Identifier names are usually derived from and can be found in the endpoint + identifier module itself (res_pjsip_endpoint_identifier_*). + You can use the CLI command "pjsip show identifiers" to see the + identifiers currently available. + One of the identifiers is "auth_username" which matches on the username in an Authentication header. This method has some security considerations because an @@ -1488,17 +1487,17 @@ how many unmatched requests are received from a single ip address before a security event is generated using the unidentified_request parameters. - + When Asterisk generates an outgoing SIP request, the From header username will be - set to this value if there is no better option (such as CallerID) to be - used. + set to this value if there is no better option (such as CallerID) to be + used. When Asterisk generates an challenge, the digest will be - set to this value if there is no better option (such as auth/realm) to be - used. + set to this value if there is no better option (such as auth/realm) to be + used. @@ -2066,7 +2065,7 @@ Provides a listing of all endpoints. For each endpoint an EndpointList event is raised that contains relevant attributes and status information. Once all endpoints have been listed an EndpointListComplete event is issued. - + @@ -2102,7 +2101,7 @@ IdentifyDetail. Some events may be listed multiple times if multiple objects are associated (for instance AoRs). Once all detail events have been raised a final EndpointDetailComplete event is issued. - +