SIP Resource using PJProject Endpoint The Endpoint is the primary configuration object. It contains the core SIP related options only, endpoints are NOT dialable entries of their own. Communication with another SIP device is accomplished via Addresses of Record (AoRs) which have one or more contacts associated with them. Endpoints NOT configured to use a transport will default to first transport found in pjsip.conf that matches its type. Example: An Endpoint has been configured with no transport. When it comes time to call an AoR, PJSIP will find the first transport that matches the type. A SIP URI of sip:5000@[11::33] will use the first IPv6 transport and try to send the request. If the anonymous endpoint identifier is in use an endpoint with the name "anonymous@domain" will be searched for as a last resort. If this is not found it will fall back to searching for "anonymous". If neither endpoints are found the anonymous endpoint identifier will not return an endpoint and anonymous calling will not be possible. Allow support for RFC3262 provisional ACK tags If set to no, do not support transmission of reliable provisional responses. As UAS, if an incoming request contains 100rel in the Required header, it is rejected with 420 Bad Extension. If set to required, require provisional responses to be sent and received reliably. As UAS, incoming requests without 100rel in the Supported header are rejected with 421 Extension Required. As UAC, outgoing requests will have 100rel in the Required header. If set to peer_supported, send provisional responses reliably if the request by the peer contained 100rel in the Supported or Require header. As UAS, if an incoming request contains 100rel in the Supported header, send 1xx responses reliably. If the request by the peer does not contain 100rel in the Supported and Require header, send responses normally. As UAC, outgoing requests will contain 100rel in the Supported header. If set to yes, indicate the support of reliable provisional responses and PRACK them if required by the peer. As UAS, if the incoming request contains 100rel in the Supported header but not in the Required header, send 1xx responses normally. If the incoming request contains 100rel in the Required header, send 1xx responses reliably. As UAC add 100rel to the Supported header and PRACK 1xx responses if required. Condense MWI notifications into a single NOTIFY. When enabled, aggregate_mwi condenses message waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled, individual NOTIFYs are sent for each mailbox. Media Codec(s) to allow Codec negotiation prefs for incoming offers. This is a string that describes how the codecs specified on an incoming SDP offer (pending) are reconciled with the codecs specified on an endpoint (configured) before being sent to the Asterisk core. The string actually specifies 4 name:value pair parameters separated by commas. Whitespace is ignored and they may be specified in any order. Note that this option is reserved for future functionality. Parameters: The codec list from the caller. (default) The codec list from the endpoint. Only common codecs with the preferred codecs first. (default) Use only the preferred codecs. Use only the non-preferred codecs. After the operation, keep all codecs. (default) After the operation, keep only the first codec. Allow transcoding. (default) Prevent transcoding. codec_prefs_incoming_offer = prefer: pending, operation: intersect, keep: all, transcode: allow Prefer the codecs coming from the caller. Use only the ones that are common. keeping the order of the preferred list. Keep all codecs in the result. Allow transcoding. Codec negotiation prefs for outgoing offers. This is a string that describes how the codecs specified in the topology that comes from the Asterisk core (pending) are reconciled with the codecs specified on an endpoint (configured) when sending an SDP offer. The string actually specifies 4 name:value pair parameters separated by commas. Whitespace is ignored and they may be specified in any order. Note that this option is reserved for future functionality. Parameters: The codec list from the core. (default) The codec list from the endpoint. Merge the lists with the preferred codecs first. (default) Only common codecs with the preferred codecs first. (default) Use only the preferred codecs. Use only the non-preferred codecs. After the operation, keep all codecs. (default) After the operation, keep only the first codec. Allow transcoding. (default) Prevent transcoding. codec_prefs_outgoing_offer = prefer: configured, operation: union, keep: first, transcode: prevent Prefer the codecs coming from the endpoint. Merge them with the codecs from the core keeping the order of the preferred list. Keep only the first one. No transcoding allowed. Codec negotiation prefs for incoming answers. This is a string that describes how the codecs specified in an incoming SDP answer (pending) are reconciled with the codecs specified on an endpoint (configured) when receiving an SDP answer. The string actually specifies 4 name:value pair parameters separated by commas. Whitespace is ignored and they may be specified in any order. Note that this option is reserved for future functionality. Parameters: The codec list in the received SDP answer. (default) The codec list from the endpoint. Merge the lists with the preferred codecs first. Only common codecs with the preferred codecs first. (default) Use only the preferred codecs. Use only the non-preferred codecs. After the operation, keep all codecs. (default) After the operation, keep only the first codec. The transcode parameter is ignored when processing answers. codec_prefs_incoming_answer = keep: first Use the defaults but keep oinly the first codec. Codec negotiation prefs for outgoing answers. This is a string that describes how the codecs that come from the core (pending) are reconciled with the codecs specified on an endpoint (configured) when sending an SDP answer. The string actually specifies 4 name:value pair parameters separated by commas. Whitespace is ignored and they may be specified in any order. Note that this option is reserved for future functionality. Parameters: The codec list that came from the core. (default) The codec list from the endpoint. Merge the lists with the preferred codecs first. Only common codecs with the preferred codecs first. (default) Use only the preferred codecs. Use only the non-preferred codecs. After the operation, keep all codecs. (default) After the operation, keep only the first codec. The transcode parameter is ignored when processing answers. codec_prefs_incoming_answer = keep: first Use the defaults but keep oinly the first codec. Enable RFC3578 overlap dialing support. AoR(s) to be used with the endpoint List of comma separated AoRs that the endpoint should be associated with. Authentication Object(s) associated with the endpoint This is a comma-delimited list of auth sections defined in pjsip.conf to be used to verify inbound connection attempts. Endpoints without an authentication object configured will allow connections without verification. Using the same auth section for inbound and outbound authentication is not recommended. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses. See the auth realm description for details. CallerID information for the endpoint Must be in the format Name <Number>, or only <Number>. Default privacy level Internal id_tag for the endpoint Dialplan context for inbound sessions Mitigation of direct media (re)INVITE glare This setting attempts to avoid creating INVITE glare scenarios by disabling direct media reINVITEs in one direction thereby allowing designated servers (according to this option) to initiate direct media reINVITEs without contention and significantly reducing call setup time. A more detailed description of how this option functions can be found on the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance Direct Media method type Method for setting up Direct Media between endpoints. Alias for the invite value. Accept Connected Line updates from this endpoint Send Connected Line updates to this endpoint Connected line method type Method used when updating connected line information. When set to invite, check the remote's Allow header and if UPDATE is allowed, send UPDATE instead of INVITE to avoid SDP renegotiation. If UPDATE is not Allowed, send INVITE. Alias for the invite value. If set to update, send UPDATE regardless of what the remote Allows. Determines whether media may flow directly between endpoints. Disable direct media session refreshes when NAT obstructs the media session Media Codec(s) to disallow DTMF mode This setting allows to choose the DTMF mode for endpoint communication. DTMF is sent out of band of the main audio stream. This supercedes the older RFC-2833 used within the older chan_sip. DTMF is sent as part of audio stream. 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 SIP INFO if not. IP address used in SDP for media handling At the time of SDP creation, the IP address defined here will be used as the media address for individual streams in the SDP. Be aware that the external_media_address option, set in Transport configuration, can also affect the final media address used in the SDP. Bind the RTP instance to the media_address If media_address is specified, this option causes the RTP instance to be bound to the specified ip address which causes the packets to be sent from that address. Force use of return port Enable the ICE mechanism to help traverse NAT Way(s) for the endpoint to be identified Endpoints and AORs can be identified in multiple ways. This option is a comma separated list of methods the endpoint can be identified. This option controls both how an endpoint is matched for incoming traffic and also how an AOR is determined if a registration occurs. You must list at least one method that also matches for AORs or the registration will fail. Matches the endpoint or AOR ID based on the username and domain in the From header (or To header for AORs). If an exact match on both username and domain/realm fails, the match is retried with just the username. Matches the endpoint or AOR ID based on the username and realm in the Authentication header. If an exact match on both username and domain/realm fails, the match is retried with just the username. This method of identification has some security considerations because an Authentication header is not present on the first message of a dialog when digest authentication is used. The client can't generate it until the server sends the challenge in a 401 response. Since Asterisk normally sends a security event when an incoming request can't be matched to an endpoint, using this method requires that the security event be deferred until a request is received with the Authentication header and only generated if the username doesn't result in a match. This may result in a delay before an attack is recognized. You can control how many unmatched requests are received from a single ip address before a security event is generated using the unidentified_request parameters in the "global" configuration object. Matches the endpoint based on the source IP address. This method of identification is not configured here but simply allowed by this configuration option. See the documentation for the identify configuration section for more details on this method of endpoint identification. Matches the endpoint based on a configured SIP header value. This method of identification is not configured here but simply allowed by this configuration option. See the documentation for the identify configuration section for more details on this method of endpoint identification. How redirects received from an endpoint are handled When a redirect is received from an endpoint there are multiple ways it can be handled. If this option is set to user the user portion of the redirect target is treated as an extension within the dialplan and dialed using a Local channel. If this option is set to uri_core the target URI is returned to the dialing application which dials it using the PJSIP channel driver and endpoint originally used. If this option is set to uri_pjsip the redirect occurs within chan_pjsip itself and is not exposed to the core at all. The uri_pjsip option has the benefit of being more efficient and also supporting multiple potential redirect targets. The con is that since redirection occurs within chan_pjsip redirecting information is not forwarded and redirection can not be prevented. NOTIFY the endpoint when state changes for any of the specified mailboxes Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state changes happen for any of the specified mailboxes. More than one mailbox can be specified with a comma-delimited string. app_voicemail mailboxes must be specified as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by external sources, such as through the res_mwi_external module, you must specify strings supported by the external system. For endpoints that SUBSCRIBE for MWI, use the mailboxes option in your AOR configuration. An MWI subscribe will replace sending unsolicited NOTIFYs The voicemail extension to send in the NOTIFY Message-Account header Default Music On Hold class Authentication object(s) used for outbound requests This is a comma-delimited list of auth sections defined in pjsip.conf used to respond to outbound connection authentication challenges. Using the same auth section for inbound and outbound authentication is not recommended. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses. See the auth realm description for details. Full SIP URI of the outbound proxy used to send requests Allow Contact header to be rewritten with the source IP address-port On inbound SIP messages from this endpoint, the Contact header or an appropriate Record-Route header will be changed to have the source IP address and port. This option does not affect outbound messages sent to this endpoint. This option helps servers communicate with endpoints that are behind NATs. This option also helps reuse reliable transport connections such as TCP and TLS. Allow use of IPv6 for RTP traffic Enforce that RTP must be symmetric Send the Diversion header, conveying the diversion information to the called user agent Send the History-Info header, conveying the diversion information to the called and calling user agents Send the P-Asserted-Identity header Send the Remote-Party-ID header Immediately send connected line updates on unanswered incoming calls. When enabled, immediately send 180 Ringing or 183 Progress response messages to the caller if the connected line information is updated before the call is answered. This can send a 180 Ringing response before the call has even reached the far end. The caller can start hearing ringback before the far end even gets the call. Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of this Asterisk box. When disabled, a connected line update must wait for another reason to send a message with the connected line information to the caller before the call is answered. You can trigger the sending of the information by using an appropriate dialplan application such as Ringing. Minimum session timers expiration period Minimum session timer expiration period. Time in seconds. Session timers for SIP packets Alias of always Maximum session timer expiration period Maximum session timer expiration period. Time in seconds. Explicit transport configuration to use This will force the endpoint to use the specified transport configuration to send SIP messages. You need to already know what kind of transport (UDP/TCP/IPv4/etc) the endpoint device will use. Not specifying a transport will select the first configured transport in pjsip.conf which is compatible with the URI we are trying to contact. Transport configuration is not affected by reloads. In order to change transports, a full Asterisk restart is required Accept identification information received from this endpoint This option determines whether Asterisk will accept identification from the endpoint from headers such as P-Asserted-Identity or Remote-Party-ID header. This option applies both to calls originating from the endpoint and calls originating from Asterisk. If no, the configured Caller-ID from pjsip.conf will always be used as the identity for the endpoint. Send private identification details to the endpoint. This option determines whether res_pjsip will send private identification information to the endpoint. If no, private Caller-ID information will not be forwarded to the endpoint. "Private" in this case refers to any method of restricting identification. Example: setting callerid_privacy to any prohib variation. Example: If trust_id_inbound is set to yes, the presence of a Privacy: id header in a SIP request or response would indicate the identification provided in the request is private. Must be of type 'endpoint'. Use Endpoint's requested packetization interval Determines whether res_pjsip will use and enforce usage of AVPF for this endpoint. If set to yes, res_pjsip will use the AVPF or SAVPF RTP profile for all media offers on outbound calls and media updates and will decline media offers not using the AVPF or SAVPF profile. 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. Determines whether res_pjsip will use and enforce usage of AVP, regardless of the RTP profile in use for this endpoint. If set to yes, res_pjsip will use the AVP, AVPF, SAVP, or SAVPF RTP profile for all media offers on outbound calls and media updates including those for DTLS-SRTP streams. If set to no, res_pjsip will use the respective RTP profile depending on configuration. Determines whether res_pjsip will use the media transport received in the offer SDP in the corresponding answer SDP. If set to yes, res_pjsip will use the received media transport. If set to no, res_pjsip will use the respective RTP profile depending on configuration. Determines whether res_pjsip will use and enforce usage of media encryption for this endpoint. res_pjsip will offer no encryption and allow no encryption to be setup. res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP transport should be used in conjunction with this option to prevent exposure of media encryption keys. res_pjsip will offer DTLS-SRTP setup. Determines whether encryption should be used if possible but does not terminate the session if not achieved. This option only applies if media_encryption is set to sdes or dtls. 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. Determines whether chan_pjsip will indicate ringing using inband progress. If set to yes, chan_pjsip will send a 183 Session Progress when told to indicate ringing and will immediately start sending ringing as audio. If set to no, chan_pjsip will send a 180 Ringing when told to indicate ringing and will NOT send it as audio. The numeric pickup groups for a channel. Can be set to a comma separated list of numbers or ranges between the values of 0-63 (maximum of 64 groups). The numeric pickup groups that a channel can pickup. Can be set to a comma separated list of numbers or ranges between the values of 0-63 (maximum of 64 groups). The named pickup groups for a channel. Can be set to a comma separated list of case sensitive strings limited by supported line length. The named pickup groups that a channel can pickup. Can be set to a comma separated list of case sensitive strings limited by supported line length. The number of in-use channels which will cause busy to be returned as device state When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the PJSIP channel driver will return busy as the device state instead of in use. Whether T.38 UDPTL support is enabled or not If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted and relayed. T.38 UDPTL error correction method No error correction should be used. Forward error correction should be used. Redundancy error correction should be used. T.38 UDPTL maximum datagram size This option can be set to override the maximum datagram of a remote endpoint for broken endpoints. Whether CNG tone detection is enabled This option can be set to send the session to the fax extension when a CNG tone is detected. How long into a call before fax_detect is disabled for the call The option determines how many seconds into a call before the fax_detect option is disabled for the call. Setting the value to zero disables the timeout. Whether NAT support is enabled on UDPTL sessions When enabled the UDPTL stack will send UDPTL packets to the source address of received packets. Whether IPv6 is used for UDPTL Sessions When enabled the UDPTL stack will use IPv6. Bind the UDPTL instance to the media_adress If media_address is specified, this option causes the UDPTL instance to be bound to the specified ip address which causes the packets to be sent from that address. Set which country's indications to use for channels created for this endpoint. Set the default language to use for channels created for this endpoint. Determines whether one-touch recording is allowed for this endpoint. record_on_feature record_off_feature The feature to enact when one-touch recording is turned on. When an INFO request for one-touch recording arrives with a Record header set to "on", this feature will be enabled for the channel. The feature designated here can be any built-in or dynamic feature defined in features.conf. This setting has no effect if the endpoint's one_touch_recording option is disabled one_touch_recording record_off_feature The feature to enact when one-touch recording is turned off. When an INFO request for one-touch recording arrives with a Record header set to "off", this feature will be enabled for the channel. The feature designated here can be any built-in or dynamic feature defined in features.conf. This setting has no effect if the endpoint's one_touch_recording option is disabled one_touch_recording record_on_feature Name of the RTP engine to use for channels created for this endpoint Determines whether SIP REFER transfers are allowed for this endpoint Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side String placed as the username portion of an SDP origin (o=) line. String used for the SDP session (s=) line. DSCP TOS bits for audio streams See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings DSCP TOS bits for video streams See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings Priority for audio streams See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings Priority for video streams See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings Determines if endpoint is allowed to initiate subscriptions with Asterisk. The minimum allowed expiry time for subscriptions initiated by the endpoint. Username to use in From header for requests to this endpoint. Username to use in From header for unsolicited MWI NOTIFYs to this endpoint. Domain to user in From header for requests to this endpoint. Verify that the provided peer certificate is valid This option only applies if media_encryption is set to dtls. It can be one of the following values: meaning no verification is done. meaning to verify the remote fingerprint. meaning to verify the remote certificate. meaning to verify both the remote fingerprint and certificate. Interval at which to renegotiate the TLS session and rekey the SRTP session This option only applies if media_encryption is set to dtls. If this is not set or the value provided is 0 rekeying will be disabled. Whether or not to automatically generate an ephemeral X.509 certificate If enabled, Asterisk will generate an X.509 certificate for each DTLS session. This option only applies if media_encryption is set to dtls. This option will be automatically enabled if webrtc is enabled and dtls_cert_file is not specified. Path to certificate file to present to peer This option only applies if media_encryption is set to dtls. Path to private key for certificate file This option only applies if media_encryption is set to dtls. Cipher to use for DTLS negotiation This option only applies if media_encryption is set to dtls. Many options for acceptable ciphers. See link for more: http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS Path to certificate authority certificate This option only applies if media_encryption is set to dtls. Path to a directory containing certificate authority certificates This option only applies if media_encryption is set to dtls. Whether we are willing to accept connections, connect to the other party, or both. This option only applies if media_encryption is set to dtls. res_pjsip will make a connection to the peer. res_pjsip will accept connections from the peer. res_pjsip will offer and accept connections from the peer. Type of hash to use for the DTLS fingerprint in the SDP. This option only applies if media_encryption is set to dtls. Determines whether 32 byte tags should be used instead of 80 byte tags. This option only applies if media_encryption is set to sdes or dtls. Variable set on a channel involving the endpoint. 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). Context to route incoming MESSAGE requests to. If specified, incoming MESSAGE requests will be routed to the indicated dialplan context. If no message_context is specified, then the context setting is used. An accountcode to set automatically on any channels created for this endpoint. If specified, any channel created for this endpoint will automatically have this accountcode set on it. Respond to a SIP invite with the single most preferred codec (DEPRECATED) Respond to a SIP invite with the single most preferred codec rather than advertising all joint codec capabilities. This limits the other side's codec choice to exactly what we prefer. This option has been deprecated in favor of incoming_call_offer_pref. Setting both options is unsupported. incoming_call_offer_pref Preferences for selecting codecs for an incoming call. Based on this setting, a joint list of preferred codecs between those received in an incoming SDP offer (remote), and those specified in the endpoint's "allow" parameter (local) es created and is passed to the Asterisk core. This list will consist of only those codecs found in both lists. Include all codecs in the local list that are also in the remote list preserving the local order. (default). Include only the first codec in the local list that is also in the remote list. Include all codecs in the remote list that are also in the local list preserving the remote order. Include only the first codec in the remote list that is also in the local list. Preferences for selecting codecs for an outgoing call. Based on this setting, a joint list of preferred codecs between those received from the Asterisk core (remote), and those specified in the endpoint's "allow" parameter (local) is created and is used to create the outgoing SDP offer. Include all codecs in the local list that are also in the remote list preserving the local order. Include all codecs in the local list preserving the local order. Include only the first codec in the local list. Include all codecs in the remote list that are also in the local list preserving the remote order. Include all codecs in the local list preserving the remote order. (default) Include only the first codec in the remote list that is also in the local list. Number of seconds between RTP comfort noise keepalive packets. At the specified interval, Asterisk will send an RTP comfort noise frame. This may be useful for situations where Asterisk is behind a NAT or firewall and must keep a hole open in order to allow for media to arrive at Asterisk. Maximum number of seconds without receiving RTP (while off hold) before terminating call. This option configures the number of seconds without RTP (while off hold) before considering a channel as dead. When the number of seconds is reached the underlying channel is hung up. By default this option is set to 0, which means do not check. Maximum number of seconds without receiving RTP (while on hold) before terminating call. This option configures the number of seconds without RTP (while on hold) before considering a channel as dead. When the number of seconds is reached the underlying channel is hung up. By default this option is set to 0, which means do not check. List of IP ACL section names in acl.conf This matches sections configured in acl.conf. The value is defined as a list of comma-delimited section names. List of IP addresses to deny access from The value is a comma-delimited list of IP addresses. IP addresses may have a subnet mask appended. The subnet mask may be written in either CIDR or dotted-decimal notation. Separate the IP address and subnet mask with a slash ('/') List of IP addresses to permit access from The value is a comma-delimited list of IP addresses. IP addresses may have a subnet mask appended. The subnet mask may be written in either CIDR or dotted-decimal notation. Separate the IP address and subnet mask with a slash ('/') List of Contact ACL section names in acl.conf This matches sections configured in acl.conf. The value is defined as a list of comma-delimited section names. List of Contact header addresses to deny The value is a comma-delimited list of IP addresses. IP addresses may have a subnet mask appended. The subnet mask may be written in either CIDR or dotted-decimal notation. Separate the IP address and subnet mask with a slash ('/') List of Contact header addresses to permit The value is a comma-delimited list of IP addresses. IP addresses may have a subnet mask appended. The subnet mask may be written in either CIDR or dotted-decimal notation. Separate the IP address and subnet mask with a slash ('/') Context for incoming MESSAGE requests. If specified, incoming SUBSCRIBE requests will be searched for the matching extension in the indicated context. If no subscribe_context is specified, then the context setting is used. Force the user on the outgoing Contact header to this value. On outbound requests, force the user portion of the Contact header to this value. Allow the sending and receiving RTP codec to differ When set to "yes" the codec in use for sending will be allowed to differ from that of the received one. PJSIP will not automatically switch the sending one to the receiving one. Enable RFC 5761 RTCP multiplexing on the RTP port With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux" attribute on all media streams. This will result in RTP and RTCP being sent and received on the same port. This shifts the demultiplexing logic to the application rather than the transport layer. This option is useful when interoperating with WebRTC endpoints since they mandate this option's use. Whether to notifies all the progress details on blind transfer Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK" after REFER has been accepted. If set to no then asterisk will not send the progress details, but immediately will send "200 OK". Whether to notifies dialog-info 'early' on InUse&Ringing state Control whether dialog-info subscriptions get 'early' state on Ringing when already INUSE. The maximum number of allowed audio streams for the endpoint This option enforces a limit on the maximum simultaneous negotiated audio streams allowed for the endpoint. The maximum number of allowed video streams for the endpoint This option enforces a limit on the maximum simultaneous negotiated video streams allowed for the endpoint. Enable RTP bundling With this option enabled, Asterisk will attempt to negotiate the use of bundle. If negotiated this will result in multiple RTP streams being carried over the same underlying transport. Note that enabling bundle will also enable the rtcp_mux option. Defaults and enables some options that are relevant to WebRTC When set to "yes" this also enables the following values that are needed in order for basic WebRTC support to work: rtcp_mux, use_avpf, ice_support, and use_received_transport. The following configuration settings also get defaulted as follows: media_encryption=dtls dtls_auto_generate_cert=yes (if dtls_cert_file is not set) dtls_verify=fingerprint dtls_setup=actpass Mailbox name to use when incoming MWI NOTIFYs are received If an MWI NOTIFY is received from this endpoint, this mailbox will be used when notifying other modules of MWI status changes. If not set, incoming MWI NOTIFYs are ignored. Follow SDP forked media when To tag is different On outgoing calls, if the UAS responds with different SDP attributes on subsequent 18X or 2XX responses (such as a port update) AND the To tag on the subsequent response is different than that on the previous one, follow it. This usually happens when the INVITE is forked to multiple UASs and more than one sends an SDP answer. This option must also be enabled in the system section for it to take effect here. Accept multiple SDP answers on non-100rel responses On outgoing calls, if the UAS responds with different SDP attributes on non-100rel 18X or 2XX responses (such as a port update) AND the To tag on the subsequent response is the same as that on the previous one, process the updated SDP. This can happen when the UAS needs to change ports for some reason such as using a separate port for custom ringback. This option must also be enabled in the system section for it to take effect here. Suppress Q.850 Reason headers for this endpoint Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. This option allows the 'Q.850' Reason header to be suppressed. Do not forward 183 when it doesn't contain SDP Certain SS7 internetworking scenarios can result in a 183 to be generated for reasons other than early media. Forwarding this 183 can cause loss of ringback tone. This flag emulates the behavior of chan_sip and prevents these 183 responses from being forwarded. Enable STIR/SHAKEN support on this endpoint Enable STIR/SHAKEN support on this endpoint. On incoming INVITEs, the Identity header will be checked for validity. On outgoing INVITEs, an Identity header will be added. STIR/SHAKEN profile containing additional configuration options A STIR/SHAKEN profile that is defined in stir_shaken.conf. Contains several options and rules used for STIR/SHAKEN. Skip authentication when receiving OPTIONS requests RFC 3261 says that the response to an OPTIONS request MUST be the same had the request been an INVITE. Some UAs use OPTIONS requests like a 'ping' and the expectation is that they will return a 200 OK. Enabling allow_unauthenticated_options will skip authentication of OPTIONS requests for the given endpoint. There are security implications to enabling this setting as it can allow information disclosure to occur - specifically, if enabled, an external party could enumerate and find the endpoint name by sending OPTIONS requests and examining the responses. The kind of security agreement negotiation to use. Currently, only mediasec is supported. List of security mechanisms supported. This is a comma-delimited list of security mechanisms to use. Each security mechanism must be in the form defined by RFC 3329 section 2.2. Geolocation profile to apply to incoming calls This geolocation profile will be applied to all calls received by the channel driver from the remote endpoint before they're forwarded to the dialplan. Geolocation profile to apply to outgoing calls This geolocation profile will be applied to all calls received by the channel driver from the dialplan before they're forwarded the remote endpoint. Authentication type Authentication objects hold the authentication information for use by other objects such as endpoints or registrations. This also allows for multiple objects to use a single auth object. See the auth_type config option for password style choices. Authentication type This option specifies which of the password style config options should be read when trying to authenticate an endpoint inbound request. If set to userpass then we'll read from the 'password' option. For md5 we'll read from 'md5_cred'. If set to google_oauth then we'll read from the refresh_token/oauth_clientid/oauth_secret fields. The following values are valid: This setting only describes whether the password is in plain text or has been pre-hashed with MD5. It doesn't describe the acceptable digest algorithms we'll accept in a received challenge. Lifetime of a nonce associated with this authentication config. MD5 Hash used for authentication. Only used when auth_type is md5. As an alternative to specifying a plain text password, you can hash the username, realm and password together one time and place the hash value here. The input to the hash function must be in the following format: <username>:<realm>:<password> For incoming authentication (asterisk is the server), the realm must match either the realm set in this object or the default_realm set in in the global object. For outgoing authentication (asterisk is the UAC), the realm must match what the server will be sending in their WWW-Authenticate header. It can't be blank unless you expect the server to be sending a blank realm in the header. You can't use pre-hashed passwords with a wildcard auth object. You can generate the hash with the following shell command: $ echo -n "myname:myrealm:mypassword" | md5sum Note the '-n'. You don't want a newline to be part of the hash. Plain text password used for authentication. Only used when auth_type is userpass. OAuth 2.0 refresh token OAuth 2.0 application's client id OAuth 2.0 application's secret SIP realm for endpoint For incoming authentication (asterisk is the UAS), this is the realm to be sent on WWW-Authenticate headers. If not specified, the global object's default_realm will be used. For outgoing authentication (asterisk is the UAC), this must either be the realm the server is expected to send, or left blank or contain a single '*' to automatically use the realm sent by the server. If you have multiple auth objects for an endpoint, the realm is also used to match the auth object to the realm the server sent. Using the same auth section for inbound and outbound authentication is not recommended. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses. If more than one auth object with the same realm or more than one wildcard auth object associated to an endpoint, we can only use the first one of each defined on the endpoint. Must be 'auth' Username to use for account Domain Alias Signifies that a domain is an alias. If the domain on a session is not found to match an AoR then this object is used to see if we have an alias for the AoR to which the endpoint is binding. This objects name as defined in configuration should be the domain alias and a config option is provided to specify the domain to be aliased. Must be of type 'domain_alias'. Domain to be aliased SIP Transport Transports There are different transports and protocol derivatives supported by res_pjsip. They are in order of preference: UDP, TCP, and WebSocket (WS). Changes to transport configuration in pjsip.conf will only be effected on a complete restart of Asterisk. A module reload will not suffice. Number of simultaneous Asynchronous Operations, can no longer be set, always set to 1 IP Address and optional port to bind to for this transport File containing a list of certificates to read (TLS ONLY, not WSS) Path to directory containing a list of certificates to read (TLS ONLY, not WSS) Certificate file for endpoint (TLS ONLY, not WSS) A path to a .crt or .pem file can be provided. However, only the certificate is read from the file, not the private key. The priv_key_file option must supply a matching key file. The certificate file can be reloaded if the filename in configuration remains unchanged. Preferred cryptography cipher names (TLS ONLY, not WSS) Comma separated list of cipher names or numeric equivalents. Numeric equivalents can be either decimal or hexadecimal (0xX). There are many cipher names. Use the CLI command pjsip list ciphers to see a list of cipher names available for your installation. See link for more: http://www.openssl.org/docs/apps/ciphers.html#CIPHER_SUITE_NAMES Domain the transport comes from External IP address to use in RTP handling When a request or response is sent out, if the destination of the message is outside the IP network defined in the option localnet, and the media address in the SDP is within the localnet network, then the media address in the SDP will be rewritten to the value defined for external_media_address. External address for SIP signalling External port for SIP signalling Method of SSL transport (TLS ONLY, not WSS) The default as defined by PJSIP. This is currently TLSv1, but may change with future releases. This option is equivalent to setting 'default' Network to consider local (used for NAT purposes). This must be in CIDR or dotted decimal format with the IP and mask separated with a slash ('/'). Password required for transport Private key file (TLS ONLY, not WSS) A path to a key file can be provided. The private key file can be reloaded if the filename in configuration remains unchanged. Protocol to use for SIP traffic Require client certificate (TLS ONLY, not WSS) Must be of type 'transport'. Require verification of client certificate (TLS ONLY, not WSS) Require verification of server certificate (TLS ONLY, not WSS) Enable TOS for the signalling sent over this transport See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information on this parameter. This option does not apply to the ws or the wss protocols. Enable COS for the signalling sent over this transport See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information on this parameter. This option does not apply to the ws or the wss protocols. The timeout (in milliseconds) to set on WebSocket connections. If a websocket connection accepts input slowly, the timeout for writes to it can be increased to keep it from being disconnected. Value is in milliseconds. Allow this transport to be reloaded. Allow this transport to be reloaded when res_pjsip is reloaded. This option defaults to "no" because reloading a transport may disrupt in-progress calls. Allow use of wildcards in certificates (TLS ONLY) In combination with verify_server, when enabled allow use of wildcards, i.e. '*.' in certs for common,and subject alt names of type DNS for TLS transport types. Names must start with the wildcard. Partial wildcards, e.g. 'f*.example.com' and 'foo.*.com' are not allowed. As well, names only match against a single level meaning '*.example.com' matches 'foo.example.com', but not 'foo.bar.example.com'. Use the same transport for outgoing requests as incoming ones. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. A way of creating an aliased name to a SIP URI Contacts are a way to hide SIP URIs from the dialplan directly. They are also used to make a group of contactable parties when in use with AoR lists. Must be of type 'contact'. SIP URI to contact peer Time to keep alive a contact Time to keep alive a contact. String style specification. Interval at which to qualify a contact Interval between attempts to qualify the contact for reachability. If 0 never qualify. Time in seconds. Timeout for qualify If the contact doesn't respond to the OPTIONS request before the timeout, the contact is marked unavailable. If 0 no timeout. Time in fractional seconds. Authenticates a qualify challenge response if needed If true and a qualify request receives a challenge response then authentication is attempted before declaring the contact available. This option does nothing as we will always complete the challenge response authentication if the qualify request is challenged. Outbound proxy used when sending OPTIONS request If set the provided URI will be used as the outbound proxy when an OPTIONS request is sent to a contact for qualify purposes. Stored Path vector for use in Route headers on outgoing requests. User-Agent header from registration. The User-Agent is automatically stored based on data present in incoming SIP REGISTER requests and is not intended to be configured manually. Endpoint name The name of the endpoint this contact belongs to Asterisk Server name Asterisk Server name on which SIP endpoint registered. IP-address of the last Via header from registration. The last Via header should contain the address of UA which sent the request. The IP-address of the last Via header is automatically stored based on data present in incoming SIP REGISTER requests and is not intended to be configured manually. IP-port of the last Via header from registration. The IP-port of the last Via header is automatically stored based on data present in incoming SIP REGISTER requests and is not intended to be configured manually. Call-ID header from registration. The Call-ID header is automatically stored based on data present in incoming SIP REGISTER requests and is not intended to be configured manually. A contact that cannot survive a restart/boot. The option is set if the incoming SIP REGISTER contact is rewritten on a reliable transport and is not intended to be configured manually. The configuration for a location of an endpoint An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no AoRs are specified, an endpoint will not be reachable by Asterisk. Beyond that, an AoR has other uses within Asterisk, such as inbound registration. An AoR is a way to allow dialing a group of Contacts that all use the same endpoint for calls. This can be used as another way of grouping a list of contacts to dial rather than specifying them each directly when dialing via the dialplan. This must be used in conjunction with the PJSIP_DIAL_CONTACTS. Registrations: For Asterisk to match an inbound registration to an endpoint, the AoR object name must match the user portion of the SIP URI in the "To:" header of the inbound SIP registration. That will usually be equivalent to the "user name" set in your hard or soft phones configuration. Permanent contacts assigned to AoR Contacts specified will be called whenever referenced by chan_pjsip. Use a separate "contact=" entry for each contact required. Contacts are specified using a SIP URI. Default expiration time in seconds for contacts that are dynamically bound to an AoR. Allow subscriptions for the specified mailbox(es) This option applies when an external entity subscribes to an AoR for Message Waiting Indications. The mailboxes specified will be subscribed to. More than one mailbox can be specified with a comma-delimited string. app_voicemail mailboxes must be specified as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by external sources, such as through the res_mwi_external module, you must specify strings supported by the external system. For endpoints that cannot SUBSCRIBE for MWI, you can set the mailboxes option in your endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint. The voicemail extension to send in the NOTIFY Message-Account header Maximum time to keep an AoR Maximum time to keep a peer with explicit expiration. Time in seconds. Maximum number of contacts that can bind to an AoR Maximum number of contacts that can associate with this AoR. This value does not affect the number of contacts that can be added with the "contact" option. It only limits contacts added through external interaction, such as registration. The rewrite_contact option registers the source address as the contact address to help with NAT and reusing connection oriented transports such as TCP and TLS. Unfortunately, refreshing a registration may register a different contact address and exceed max_contacts. The remove_existing and remove_unavailable options can help by removing either the soonest to expire or unavailable contact(s) over max_contacts which is likely the old rewrite_contact contact source address being refreshed. This should be set to 1 and remove_existing set to yes if you wish to stick with the older chan_sip behaviour. Minimum keep alive time for an AoR Minimum time to keep a peer with an explicit expiration. Time in seconds. Determines whether new contacts replace existing ones. On receiving a new registration to the AoR should it remove enough existing contacts not added or updated by the registration to satisfy max_contacts? Any removed contacts will expire the soonest. The rewrite_contact option registers the source address as the contact address to help with NAT and reusing connection oriented transports such as TCP and TLS. Unfortunately, refreshing a registration may register a different contact address and exceed max_contacts. The remove_existing option can help by removing the soonest to expire contact(s) over max_contacts which is likely the old rewrite_contact contact source address being refreshed. This should be set to yes and max_contacts set to 1 if you wish to stick with the older chan_sip behaviour. Determines whether new contacts should replace unavailable ones. The effect of this setting depends on the setting of remove_existing. If remove_existing is set to no (default), setting remove_unavailable to yes will remove only unavailable contacts that exceed max_contacts to allow an incoming REGISTER to complete sucessfully. If remove_existing is set to yes, setting remove_unavailable to yes will prioritize unavailable contacts for removal instead of just removing the contact that expires the soonest. See remove_existing and max_contacts for further information about how these 3 settings interact. Must be of type 'aor'. Interval at which to qualify an AoR Interval between attempts to qualify the AoR for reachability. If 0 never qualify. Time in seconds. Timeout for qualify If the contact doesn't respond to the OPTIONS request before the timeout, the contact is marked unavailable. If 0 no timeout. Time in fractional seconds. Authenticates a qualify challenge response if needed If true and a qualify request receives a challenge response then authentication is attempted before declaring the contact available. This option does nothing as we will always complete the challenge response authentication if the qualify request is challenged. Outbound proxy used when sending OPTIONS request If set the provided URI will be used as the outbound proxy when an OPTIONS request is sent to a contact for qualify purposes. Enables Path support for REGISTER requests and Route support for other requests. When this option is enabled, the Path headers in register requests will be saved and its contents will be used in Route headers for outbound out-of-dialog requests and in Path headers for outbound 200 responses. Path support will also be indicated in the Supported header. Options that apply to the SIP stack as well as other system-wide settings The settings in this section are global. In addition to being global, the values will not be re-evaluated when a reload is performed. This is because the values must be set before the SIP stack is initialized. The only way to reset these values is to either restart Asterisk, or unload res_pjsip.so and then load it again. Set transaction timer T1 value (milliseconds). Timer T1 is the base for determining how long to wait before retransmitting requests that receive no response when using an unreliable transport (e.g. UDP). For more information on this timer, see RFC 3261, Section 17.1.1.1. Set transaction timer B value (milliseconds). Timer B determines the maximum amount of time to wait after sending an INVITE request before terminating the transaction. It is recommended that this be set to 64 * Timer T1, but it may be set higher if desired. For more information on this timer, see RFC 3261, Section 17.1.1.1. Use the short forms of common SIP header names. Initial number of threads in the res_pjsip threadpool. The amount by which the number of threads is incremented when necessary. Number of seconds before an idle thread should be disposed of. Maximum number of threads in the res_pjsip threadpool. A value of 0 indicates no maximum. Disable automatic switching from UDP to TCP transports. Disable automatic switching from UDP to TCP transports if outgoing request is too large. See RFC 3261 section 18.1.1. Follow SDP forked media when To tag is different On outgoing calls, if the UAS responds with different SDP attributes on subsequent 18X or 2XX responses (such as a port update) AND the To tag on the subsequent response is different than that on the previous one, follow it. This option must also be enabled on endpoints that require this functionality. Follow SDP forked media when To tag is the same On outgoing calls, if the UAS responds with different SDP attributes on non-100rel 18X or 2XX responses (such as a port update) AND the To tag on the subsequent response is the same as that on the previous one, process the updated SDP. This option must also be enabled on endpoints that require this functionality. Disable the use of rport in outgoing requests. Remove "rport" parameter from the outgoing requests. Must be of type 'system' UNLESS the object name is 'system'. Options that apply globally to all SIP communications The settings in this section are global. Unlike options in the system section, these options can be refreshed by performing a reload. Value used in Max-Forwards header for SIP requests. The interval (in seconds) to send keepalives to active connection-oriented transports. The interval (in seconds) to check for expired contacts. Disable Multi Domain support If disabled it can improve realtime performance by reducing the number of database requests. The maximum amount of time from startup that qualifies should be attempted on all contacts. If greater than the qualify_frequency for an aor, qualify_frequency will be used instead. The number of seconds over which to accumulate unidentified requests. If unidentified_request_count unidentified requests are received during unidentified_request_period, a security event will be generated. The number of unidentified requests from a single IP to allow. If unidentified_request_count unidentified requests are received during unidentified_request_period, a security event will be generated. The interval at which unidentified requests are older than twice the unidentified_request_period are pruned. Must be of type 'global' UNLESS the object name is 'global'. 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. Endpoint to use when sending an outbound request to a URI without a specified endpoint. The voicemail extension to send in the NOTIFY Message-Account header if not specified on endpoint or aor Enable/Disable SIP debug logging. Valid options include yes, no, or 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. One of the identifiers is "auth_username" which matches on the username in an Authentication header. This method has some security considerations because an Authentication header is not present on the first message of a dialog when digest authentication is used. The client can't generate it until the server sends the challenge in a 401 response. Since Asterisk normally sends a security event when an incoming request can't be matched to an endpoint, using auth_username requires that the security event be deferred until a request is received with the Authentication header and only generated if the username doesn't result in a match. This may result in a delay before an attack is recognized. You can control 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. When Asterisk generates a challenge, the digest realm will be set to this value if there is no better option (such as auth/realm) to be used. MWI taskprocessor high water alert trigger level. On a heavily loaded system you may need to adjust the taskprocessor queue limits. If any taskprocessor queue size reaches its high water level then pjsip will stop processing new requests until the alert is cleared. The alert clears when all alerting taskprocessor queues have dropped to their low water clear level. MWI taskprocessor low water clear alert level. On a heavily loaded system you may need to adjust the taskprocessor queue limits. If any taskprocessor queue size reaches its high water level then pjsip will stop processing new requests until the alert is cleared. The alert clears when all alerting taskprocessor queues have dropped to their low water clear level. Set to -1 for the low water level to be 90% of the high water level. Enable/Disable sending unsolicited MWI to all endpoints on startup. When the initial unsolicited MWI notification are enabled on startup then the initial notifications get sent at startup. If you have a lot of endpoints (thousands) that use unsolicited MWI then you may want to consider disabling the initial startup notifications. When the initial unsolicited MWI notifications are disabled on startup then the notifications will start on the endpoint's next contact update. Enable/Disable ignoring SIP URI user field options. If you have this option enabled and there are semicolons in the user field of a SIP URI then the field is truncated at the first semicolon. This effectively makes the semicolon a non-usable character for PJSIP endpoint names, extensions, and AORs. This can be useful for improving compatibility with an ITSP that likes to use user options for whatever reason. sip:1235557890;phone-context=national@x.x.x.x;user=phone 1235557890;phone-context=national 1235557890 The caller-id and redirecting number strings obtained from incoming SIP URI user fields are always truncated at the first semicolon. Place caller-id information into Contact header This option will cause Asterisk to place caller-id information into generated Contact headers. Enable sending AMI ContactStatus event when a device refreshes its registration. Trigger scope for taskprocessor overloads This option specifies the trigger the distributor will use for detecting taskprocessor overloads. When it detects an overload condition, the distrubutor will stop accepting new requests until the overload is cleared. (default) Any taskprocessor overload will trigger. Only pjsip taskprocessor overloads will trigger. No overload detection will be performed. The "none" and "pjsip_only" options should be used with extreme caution and only to mitigate specific issues. Under certain conditions they could make things worse. Advertise support for RFC4488 REFER subscription suppression Allow 180 after 183 Allow Asterisk to send 180 Ringing to an endpoint after 183 Session Progress has been send. If disabled Asterisk will instead send only a 183 Session Progress to the endpoint. (default: "no") If we should return all codecs on re-INVITE without SDP On reception of a re-INVITE without SDP Asterisk will send an SDP offer in the 200 OK response containing all configured codecs on the endpoint, instead of simply those that have already been negotiated. RFC 3261 specifies this as a SHOULD requirement.