res_pjsip: SIP Resource using PJProject¶

This configuration documentation is for functionality provided by res_pjsip.

Configuration File: pjsip.conf¶

[endpoint]: Endpoint¶

Since¶

12.0.0

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.

Configuration Option Reference¶

Configuration Option Descriptions¶

100rel¶

Since: 12.2.0

• no - 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.
  

• required - 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.
  

• peer_supported - 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.
  

• yes - 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.
  

accept_multiple_sdp_answers¶

Since: 13.22.0, 15.5.0

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.

accountcode¶

Since: 13.5.0

If specified, any channel created for this endpoint will automatically have this accountcode set on it.

acl¶

Since: 13.10.0

This matches sections configured in 'acl.conf'. The value is defined as a list of comma-delimited section names.

aggregate_mwi¶

Since: 12.0.0

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.

allow_unauthenticated_options¶

Since: 16.18.0, 18.4.0

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.

aors¶

Since: 12.0.0

List of comma separated AoRs that the endpoint should be associated with.

asymmetric_rtp_codec¶

Since: 13.13.0, 14.2.0

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.

auth¶

Since: 12.2.0

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.

bind_rtp_to_media_address¶

Since: 13.8.0

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.

bundle¶

Since: 15.0.0

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.

call_group¶

Since: 12.2.0

Can be set to a comma separated list of numbers or ranges between the values of 0-63 (maximum of 64 groups).

callerid¶

Since: 12.2.0

Must be in the format 'Name ', or only ''.

callerid_privacy¶

Since: 12.7.0

• allowed_not_screened

• allowed_passed_screen

• allowed_failed_screen

• allowed

• prohib_not_screened

• prohib_passed_screen

• prohib_failed_screen

• prohib

• unavailable

codec_prefs_incoming_answer¶

Since: 18.0.0

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:

• prefer: < pending | configured > -
  

  • pending - The codec list in the received SDP answer. (default)
    

  • configured - The codec list from the endpoint.
    

• operation : < union | intersect | only_preferred | only_nonpreferred > -
  

  • union - Merge the lists with the preferred codecs first.
    

  • intersect - Only common codecs with the preferred codecs first. (default)
    

  • only_preferred - Use only the preferred codecs.
    

  • only_nonpreferred - Use only the non-preferred codecs.
    

• keep : < all | first > -
  

  • all - After the operation, keep all codecs. (default)
    

  • first - After the operation, keep only the first codec.
    

• transcode : < allow | prevent > - The transcode parameter is ignored when processing answers.
  
  

codec_prefs_incoming_answer = keep: first

codec_prefs_incoming_offer¶

Since: 18.0.0

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:

• prefer: < pending | configured > -
  

  • pending - The codec list from the caller. (default)
    

  • configured - The codec list from the endpoint.
    

• operation : < intersect | only_preferred | only_nonpreferred > -
  

  • intersect - Only common codecs with the preferred codecs first. (default)
    

  • only_preferred - Use only the preferred codecs.
    

  • only_nonpreferred - Use only the non-preferred codecs.
    

• keep : < all | first > -
  

  • all - After the operation, keep all codecs. (default)
    

  • first - After the operation, keep only the first codec.
    

• transcode : < allow | prevent > -
  

  • allow - Allow transcoding. (default)
    

  • prevent - Prevent transcoding.
    
    

codec_prefs_incoming_offer = prefer: pending, operation: intersect, keep: all, transcode: allow

codec_prefs_outgoing_answer¶

Since: 18.0.0

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:

• prefer: < pending | configured > -
  

  • pending - The codec list that came from the core. (default)
    

  • configured - The codec list from the endpoint.
    

• operation : < union | intersect | only_preferred | only_nonpreferred > -
  

  • union - Merge the lists with the preferred codecs first.
    

  • intersect - Only common codecs with the preferred codecs first. (default)
    

  • only_preferred - Use only the preferred codecs.
    

  • only_nonpreferred - Use only the non-preferred codecs.
    

• keep : < all | first > -
  

  • all - After the operation, keep all codecs. (default)
    

  • first - After the operation, keep only the first codec.
    

• transcode : < allow | prevent > - The transcode parameter is ignored when processing answers.
  
  

codec_prefs_incoming_answer = keep: first

codec_prefs_outgoing_offer¶

Since: 18.0.0

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:

• prefer: < pending | configured > -
  

  • pending - The codec list from the core. (default)
    

  • configured - The codec list from the endpoint.
    

• operation : < union | intersect | only_preferred | only_nonpreferred > -
  

  • union - Merge the lists with the preferred codecs first. (default)
    

  • intersect - Only common codecs with the preferred codecs first. (default)
    

  • only_preferred - Use only the preferred codecs.
    

  • only_nonpreferred - Use only the non-preferred codecs.
    

• keep : < all | first > -
  

  • all - After the operation, keep all codecs. (default)
    

  • first - After the operation, keep only the first codec.
    

• transcode : < allow | prevent > -
  

  • allow - Allow transcoding. (default)
    

  • prevent - Prevent transcoding.
    
    

codec_prefs_outgoing_offer = prefer: configured, operation: union, keep: first, transcode: prevent

connected_line_method¶

Since: 12.2.0

Method used when updating connected line information.

• invite - 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.
  

• reinvite - Alias for the 'invite' value.
  

• update - If set to 'update', send UPDATE regardless of what the remote Allows.
  

contact_acl¶

Since: 13.10.0

This matches sections configured in 'acl.conf'. The value is defined as a list of comma-delimited section names.

contact_deny¶

Since: 13.10.0

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 ('/')

contact_permit¶

Since: 13.10.0

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 ('/')

contact_user¶

Since: 13.12.0, 14.1.0

On outbound requests, force the user portion of the Contact header to this value.

cos_audio¶

Since: 12.0.0

See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for more information about QoS settings

cos_video¶

Since: 12.0.0

See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for more information about QoS settings

deny¶

Since: 13.10.0

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 ('/')

device_state_busy_at¶

Since: 12.0.0

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.

direct_media_glare_mitigation¶

Since: 12.2.0

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 in the Asterisk documentation https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Concepts/SIP-Direct-Media-Reinvite-Glare-Avoidance/

• none

• outgoing

• incoming

direct_media_method¶

Since: 12.2.0

Method for setting up Direct Media between endpoints.

• invite

• reinvite - Alias for the 'invite' value.
  

• update

dtls_auto_generate_cert¶

Since: 15.2.0

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.

dtls_ca_file¶

Since: 12.2.0

This option only applies if media_encryption is set to 'dtls'.

dtls_ca_path¶

Since: 12.2.0

This option only applies if media_encryption is set to 'dtls'.

dtls_cert_file¶

Since: 12.2.0

This option only applies if media_encryption is set to 'dtls'.

dtls_cipher¶

Since: 12.2.0

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

dtls_fingerprint¶

Since: 12.7.0

This option only applies if media_encryption is set to 'dtls'.

• SHA-256

• SHA-1

dtls_private_key¶

Since: 12.2.0

This option only applies if media_encryption is set to 'dtls'.

dtls_rekey¶

Since: 12.7.0

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.

dtls_setup¶

Since: 12.2.0

This option only applies if media_encryption is set to 'dtls'.

• active - res_pjsip will make a connection to the peer.
  

• passive - res_pjsip will accept connections from the peer.
  

• actpass - res_pjsip will offer and accept connections from the peer.
  

dtls_verify¶

Since: 12.7.0

This option only applies if media_encryption is set to 'dtls'.

It can be one of the following values:

• no - meaning no verification is done.
  

• fingerprint - meaning to verify the remote fingerprint.
  

• certificate - meaning to verify the remote certificate.
  

• yes - meaning to verify both the remote fingerprint and certificate.
  

dtmf_mode¶

Since: 12.2.0

This setting allows to choose the DTMF mode for endpoint communication.

• rfc4733 - DTMF is sent out of band of the main audio stream. This supercedes the older RFC-2833 used within the older 'chan_sip'.
  

• inband - DTMF is sent as part of audio stream.
  

• info - DTMF is sent as SIP INFO packets.
  

• auto - DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.
  

• auto_info - DTMF is sent as RFC 4733 if the other side supports it or as SIP INFO if not.
  

fax_detect¶

Since: 12.0.0

This option can be set to send the session to the fax extension when a CNG tone is detected.

fax_detect_timeout¶

Since: 13.11.0

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.

follow_early_media_fork¶

Since: 13.22.0, 15.5.0

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.

force_avp¶

Since: 12.4.0

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.

g726_non_standard¶

Since: 13.5.0

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.

geoloc_incoming_call_profile¶

Since: 16.28.0, 18.14.0, 19.6.0

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.

geoloc_outgoing_call_profile¶

Since: 16.28.0, 18.14.0, 19.6.0

This geolocation profile will be applied to all calls received by the channel driver from the dialplan before they're forwarded the remote endpoint.

identify_by¶

Since: 13.19.0, 15.2.0

Endpoints and AORs can be identified in multiple ways. This option is a comma separated list of methods the endpoint can be identified.

• username - 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.
  

• auth_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.
  

  Note

  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.

• ip - 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.
  

• header - 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.
  

• request_uri - Matches the endpoint based on the configured SIP request uri.
  This method of identification is not configured here but simply allowed by this configuration option.
  

ignore_183_without_sdp¶

Since: 13.26.0, 16.3.0

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.

inband_progress¶

Since: 12.0.0

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.

incoming_call_offer_pref¶

Since: 18.0.0

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.

• local - Include all codecs in the local list that are also in the remote list preserving the local order. (default).
  

• local_first - Include only the first codec in the local list that is also in the remote list.
  

• remote - Include all codecs in the remote list that are also in the local list preserving the remote order.
  

• remote_first - Include only the first codec in the remote list that is also in the local list.
  

incoming_mwi_mailbox¶

Since: 13.18.0, 14.7.0, 15.1.0

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.

mailboxes¶

Since: 12.0.0

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.

max_audio_streams¶

Since: 15.0.0

This option enforces a limit on the maximum simultaneous negotiated audio streams allowed for the endpoint.

max_video_streams¶

Since: 15.0.0

This option enforces a limit on the maximum simultaneous negotiated video streams allowed for the endpoint.

media_address¶

Since: 18.22.0, 20.7.0, 21.2.0

At the time of SDP creation, the IP address defined here will be used as the media address for individual streams in the SDP.

media_encryption¶

Since: 12.2.0

• no - res_pjsip will offer no encryption and allow no encryption to be setup.
  

• sdes - 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.
  

• dtls - res_pjsip will offer DTLS-SRTP setup.
  

media_encryption_optimistic¶

Since: 13.1.0

This option only applies if media_encryption is set to 'sdes' or 'dtls'.

media_use_received_transport¶

Since: 12.4.0

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.

message_context¶

Since: 13.5.0

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.

named_call_group¶

Since: 12.2.0

Can be set to a comma separated list of case sensitive strings limited by supported line length.

named_pickup_group¶

Since: 12.2.0

Can be set to a comma separated list of case sensitive strings limited by supported line length.

notify_early_inuse_ringing¶

Since: 13.17.0, 14.6.0

Control whether dialog-info subscriptions get 'early' state on Ringing when already INUSE.

outbound_auth¶

Since: 12.2.0

This is a comma-delimited list of auth sections defined in pjsip.conf used to respond to outbound connection authentication challenges.

outgoing_call_offer_pref¶

Since: 18.0.0

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.

• local - Include all codecs in the local list that are also in the remote list preserving the local order.
  

• local_merge - Include all codecs in the local list preserving the local order.
  

• local_first - Include only the first codec in the local list.
  

• remote - Include all codecs in the remote list that are also in the local list preserving the remote order.
  

• remote_merge - Include all codecs in the local list preserving the remote order. (default)
  

• remote_first - Include only the first codec in the remote list that is also in the local list.
  

overlap_context¶

Since: 18.17.0, 20.2.0

Dialplan context to use for overlap dialing extension matching. If not specified, the context configured for the endpoint will be used. If specified, the extensions/patterns in the specified context will be used for determining if a full number has been received from the endpoint.

permit¶

Since: 13.10.0

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 ('/')

pickup_group¶

Since: 12.2.0

Can be set to a comma separated list of numbers or ranges between the values of 0-63 (maximum of 64 groups).

preferred_codec_only¶

Since: 15.0.0

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.

record_off_feature¶

Since: 12.0.0

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.

record_on_feature¶

Since: 12.0.0

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.

redirect_method¶

Since: 12.2.0

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.

• user

• uri_core

• uri_pjsip

refer_blind_progress¶

Since: 13.17.0, 14.6.0

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".

rewrite_contact¶

Since: 12.0.0

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.

rpid_immediate¶

Since: 13.4.0

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.

rtcp_mux¶

Since: 13.15.0, 14.4.0

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.

rtp_keepalive¶

Since: 13.5.0

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.

rtp_timeout¶

Since: 13.5.0

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.

rtp_timeout_hold¶

Since: 13.5.0

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.

security_mechanisms¶

Since: 21.0.0

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.

security_negotiation¶

Since: 21.0.0

• no

• mediasec

set_var¶

Since: 12.2.0

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).

srtp_tag_32¶

Since: 12.0.0

This option only applies if media_encryption is set to 'sdes' or 'dtls'.

stir_shaken¶

Since: 18.22.0, 20.7.0, 21.2.0

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¶

Since: 16.26.0, 18.12.0, 19.4.0

A STIR/SHAKEN profile that is defined in stir_shaken.conf. Contains several options and rules used for STIR/SHAKEN.

subscribe_context¶

Since: 13.11.0

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.

suppress_moh_on_sendonly¶

Since: 20.11.0, 21.6.0, 22.1.0

Normally, when one party in a call sends Asterisk an SDP with a "sendonly" or "inactive" attribute it means "hold" and causes Asterisk to start playing MOH back to the other party. This can be problematic if it happens at certain times, such as in a 183 Progress message, because the MOH will replace any early media you may be playing to the calling party. If you set this option to "yes" on an endpoint and the endpoint receives an SDP with "sendonly" or "inactive", Asterisk will NOT play MOH back to the other party.

suppress_q850_reason_headers¶

Since: 13.23.0, 15.6.0

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.

t38_bind_udptl_to_media_address¶

Since: 16.22.0, 18.8.0

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.

t38_udptl¶

Since: 12.0.0

If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted and relayed.

t38_udptl_ec¶

Since: 12.2.0

• none - No error correction should be used.
  

• fec - Forward error correction should be used.
  

• redundancy - Redundancy error correction should be used.
  

t38_udptl_ipv6¶

Since: 12.0.0

When enabled the UDPTL stack will use IPv6.

t38_udptl_maxdatagram¶

Since: 12.0.0

This option can be set to override the maximum datagram of a remote endpoint for broken endpoints.

t38_udptl_nat¶

Since: 12.0.0

When enabled the UDPTL stack will send UDPTL packets to the source address of received packets.

tenantid¶

Since: 18.25.0, 20.10.0, 21.5.0

Sets the tenant ID for this endpoint. When a channel is created, tenantid will be set to this value. It can be changed via dialplan later if needed.

timers¶

Since: 12.2.0

• no

• yes

• required

• always

• forced - Alias of always
  

timers_min_se¶

Since: 12.0.0

Minimum session timer expiration period. Time in seconds.

timers_sess_expires¶

Since: 12.0.0

Maximum session timer expiration period. Time in seconds.

tos_audio¶

Since: 12.2.0

See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for more information about QoS settings

tos_video¶

Since: 12.2.0

See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for more information about QoS settings

transport¶

Since: 12.0.0

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.

trust_id_inbound¶

Since: 12.0.0

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.

trust_id_outbound¶

Since: 12.0.0

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.

use_avpf¶

Since: 12.0.0

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.

webrtc¶

Since: 15.0.0

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

[auth]: Authentication type¶

Since¶

12.0.0

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 security mechanism choices.

Configuration Option Reference¶

Configuration Option Descriptions¶

auth_type¶

Since: 12.0.0

If set to 'google_oauth' then we'll read from the refresh_token/oauth_clientid/oauth_secret parameters. If set to 'digest' then we'll read from the 'password' and/or 'password_digest' parameters. The older 'md5' and 'userpass' values are deprecated and converted to 'digest'.

• userpass - Deprecated. Use 'digest'.
  

• md5 - Deprecated. Use 'digest'.
  

• google_oauth - If selected, the 'refresh_token', 'oauth_clientid' and 'oauth_secret' parameters must be provided.
  

• digest - If selected, the 'password' and/or one or more 'password_digest' parameters must be provided.
  
  

md5_cred¶

Since: 12.0.0

Use the 'password_digest' parameter instead. If supplied, a 'password_digest' parameter will be created for it.

password¶

Since: 12.0.0

Only used when auth_type is 'digest'.

password_digest¶

Since: 20.12.0, 21.7.0, 22.2.0

Only used when auth_type is 'digest'. As an alternative to specifying a plain text password, you can specify one or more pre-computed digests separated by commas.

'password_digest= [,]...'

• <digest-spec> - :
  

• <hash-algorithm> - One of the supported hash algorithms which currently are
  You can see the current list by running the CLI command 'pjproject show buildopts'.
  

  • MD5 - Supported by all versions of OpenSSL and pjproject
    

  • SHA-256 - Supported by OpenSSL versions >> 1.0.0 and pjproject versions >= 2.15.1
    

  • SHA-512-256 - Supported by OpenSSL versions >= 1.1.1 and pjproject versions >= 2.15.1
    

• <hashed-credential> - The result of passing the following string through the selected hash algorithm: '::'
  You can create the hash by piping the string into the appropriate hash/checksum program. See the description for the 'realm' parameter for info on how to set it.
  

$ echo -n "myname:myrealm:mypassword" | openssl dgst -md5
MD5(stdin)= dce9ccd0a69e3ef90d8b9bf725053e78

password_digest = md5:dce9ccd0a69e3ef90d8b9bf725053e78

realm¶

Since: 12.0.0

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.

supported_algorithms_uac¶

Since: 20.12.0, 21.7.0, 22.2.0

Valid values:

• md5 - Supported by all versions of OpenSSL and pjproject
  

• sha-256 - Supported by all versions of OpenSSL but only pjproject versions > 2.14.1
  

• sha-512-256 - Supported by OpenSSL versions >= 1.1.1 and pjproject versions > 2.14.1
  The default may be specified by the 'default_auth_algorithms_uac' parameter in the global object. If that's not specified, the default is "MD5".
  

supported_algorithms_uas¶

Since: 20.12.0, 21.7.0, 22.2.0

Valid values:

• md5 - Supported by all versions of OpenSSL and pjproject
  

• sha-256 - Supported by all versions of OpenSSL but only pjproject versions > 2.14.1
  

• sha-512-256 - Supported by OpenSSL versions >= 1.1.1 and pjproject versions > 2.14.1
  The default may be specified by the 'default_auth_algorithms_uas' parameter in the global object. If that's not specified, the default is "MD5".
  

[domain_alias]: Domain Alias¶

Since¶

13.20.0, 15.3.0

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.

Configuration Option Reference¶

[transport]: SIP Transport¶

Since¶

13.8.0

Transports

There are different transports and protocol derivatives supported by 'res_pjsip'. They are in order of preference: UDP, TCP, and WebSocket (WS).

Configuration Option Reference¶

Configuration Option Descriptions¶

allow_reload¶

Since: 13.8.0

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_wildcard_certs¶

Since: 16.28.0, 18.14.0, 19.6.0

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'.

cert_file¶

Since: 13.8.0

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.

cipher¶

Since: 12.2.0

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

cos¶

Since: 12.0.0

See 'https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service' for more information on this parameter.

external_media_address¶

Since: 12.0.0

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'.

local_net¶

Since: 12.2.0

This must be in CIDR or dotted decimal format with the IP and mask separated with a slash ('/').

method¶

Since: 12.2.0

The availability of each of these options is dependent on the version and configuration of the underlying PJSIP library.

• default - The default as defined by PJSIP. This is currently TLSv1, but may change with future releases.
  

• unspecified - This option is equivalent to setting 'default'
  

• tlsv1

• tlsv1_1

• tlsv1_2

• tlsv1_3

• sslv2

• sslv3

• sslv23

priv_key_file¶

Since: 13.8.0

A path to a key file can be provided. The private key file can be reloaded if the filename in configuration remains unchanged.

protocol¶

Since: 12.2.0

• udp

• tcp

• tls

• ws

• wss

• flow

symmetric_transport¶

Since: 13.15.0, 14.4.0

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.

tcp_keepalive_enable¶

Since: 18.23.0, 20.8.0, 21.3.0

When set to 'yes', TCP keepalive messages are sent to verify that the endpoint is still reachable. This can help detect dead TCP connections in environments where connections may be silently dropped (e.g., NAT timeouts).

tcp_keepalive_idle_time¶

Since: 18.23.0, 20.8.0, 21.3.0

Specifies the amount of time in seconds that the connection must be idle before the first TCP keepalive probe is sent. An idle connection is defined as a connection in which no data has been sent or received by the application.

tcp_keepalive_interval_time¶

Since: 18.23.0, 20.8.0, 21.3.0

Specifies the interval in seconds between individual TCP keepalive probes, once the first probe is sent. This interval is used for subsequent probes if the peer does not respond to the previous probe.

tcp_keepalive_probe_count¶

Since: 18.23.0, 20.8.0, 21.3.0

Specifies the maximum number of TCP keepalive probes to send before considering the connection dead and notifying the application. If the peer does not respond after this many probes, the connection is considered broken.

tos¶

Since: 12.2.0

See 'https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service' for more information on this parameter.

websocket_write_timeout¶

Since: 11.11.0, 12.4.0

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.

[contact]: 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.

Configuration Option Reference¶

Configuration Option Descriptions¶

authenticate_qualify¶

If true and a qualify request receives a challenge response then authentication is attempted before declaring the contact available.

call_id¶

The Call-ID header is automatically stored based on data present in incoming SIP REGISTER requests and is not intended to be configured manually.

endpoint¶

The name of the endpoint this contact belongs to

expiration_time¶

Time to keep alive a contact. String style specification.

outbound_proxy¶

If set the provided URI will be used as the outbound proxy when an OPTIONS request is sent to a contact for qualify purposes.

prune_on_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.

qualify_2xx_only¶

If true only mark a contact as available if the qualify OPTIONS request receives a 2XX response.

qualify_frequency¶

Interval between attempts to qualify the contact for reachability. If '0' never qualify. Time in seconds.

qualify_timeout¶

If the contact doesn't respond to the OPTIONS request before the timeout, the contact is marked unavailable. This includes time spent performing any required DNS lookup(s) prior to sending the OPTIONS. If '0' no timeout. Time in fractional seconds.

reg_server¶

Asterisk Server name on which SIP endpoint registered.

user_agent¶

The User-Agent is automatically stored based on data present in incoming SIP REGISTER requests and is not intended to be configured manually.

via_addr¶

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.

via_port¶

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.

[aor]: The configuration for a location of an endpoint¶

Since¶

13.35.0, 16.12.0, 17.6.0

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.

Configuration Option Reference¶

Configuration Option Descriptions¶

authenticate_qualify¶

Since: 12.0.0

If true and a qualify request receives a challenge response then authentication is attempted before declaring the contact available.

contact¶

Since: 12.2.0

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.

mailboxes¶

Since: 12.0.0

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.

max_contacts¶

Since: 12.0.0

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.

maximum_expiration¶

Since: 12.0.0

Maximum time to keep a peer with explicit expiration. Time in seconds.

minimum_expiration¶

Since: 12.0.0

Minimum time to keep a peer with an explicit expiration. Time in seconds.

outbound_proxy¶

Since: 12.0.0

If set the provided URI will be used as the outbound proxy when an OPTIONS request is sent to a contact for qualify purposes.

qualify_2xx_only¶

Since: 20.12.0, 21.7.0, 22.2.0

If true only mark a contact as available if the qualify OPTIONS request receives a 2XX response.

qualify_frequency¶

Since: 12.0.0

Interval between attempts to qualify the AoR for reachability. If '0' never qualify. Time in seconds.

qualify_timeout¶

Since: 13.4.0

If the contact doesn't respond to the OPTIONS request before the timeout, the contact is marked unavailable. This includes time spent performing any required DNS lookup(s) prior to sending the OPTIONS. If '0' no timeout. Time in fractional seconds.

remove_existing¶

Since: 12.0.0

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.

remove_unavailable¶

Since: 16.22.0, 18.8.0

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.

support_path¶

Since: 12.1.0

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.

[system]: Options that apply to the SIP stack as well as other system-wide settings¶

Since¶

12.2.0

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.

Configuration Option Reference¶

Configuration Option Descriptions¶

accept_multiple_sdp_answers¶

Since: 13.22.0, 15.5.0

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.

disable_rport¶

Since: 13.35.0, 16.12.0, 17.6.0

Remove "rport" parameter from the outgoing requests.

disable_tcp_switch¶

Since: 13.1.0

Disable automatic switching from UDP to TCP transports if outgoing request is too large. See RFC 3261 section 18.1.1.

follow_early_media_fork¶

Since: 13.22.0, 15.5.0

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.

timer_b¶

Since: 12.0.0

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.

timer_t1¶

Since: 12.0.0

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.

[global]: Options that apply globally to all SIP communications¶

Since¶

12.0.0

The settings in this section are global. Unlike options in the 'system' section, these options can be refreshed by performing a reload.

Configuration Option Reference¶

Configuration Option Descriptions¶

all_codecs_on_empty_reinvite¶

Since: 16.30.0, 18.16.0, 19.8.0, 20.1.0

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.

allow_sending_180_after_183¶

Since: 16.26.0, 18.12.0, 19.4.0

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")

default_auth_algorithms_uac¶

Since: 20.12.0, 21.7.0, 22.2.0

Valid values:

• md5 - Supported by all versions of OpenSSL and pjproject
  

• sha-256 - Supported by all versions of OpenSSL but only pjproject versions > 2.14.1
  

• sha-512-256 - Supported by OpenSSL versions >= 1.1.1 and pjproject versions > 2.14.1
  If not specified, the default is 'MD5' only.
  

default_auth_algorithms_uas¶

Since: 20.12.0, 21.7.0, 22.2.0

Valid values:

• md5 - Supported by all versions of OpenSSL and pjproject
  

• sha-256 - Supported by all versions of OpenSSL but only pjproject versions > 2.14.1
  

• sha-512-256 - Supported by OpenSSL versions >= 1.1.1 and pjproject versions > 2.14.1
  If not specified, the default is 'MD5' only.
  

disable_multi_domain¶

Since: 13.12.0, 14.1.0

If disabled it can improve realtime performance by reducing the number of database requests.

endpoint_identifier_order¶

Since: 13.3.0

ignore_uri_user_options¶

Since: 13.12.0, 14.1.0

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

mwi_disable_initial_unsolicited¶

Since: 13.12.0, 14.1.0

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.

mwi_tps_queue_high¶

Since: 13.12.0, 14.1.0

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_tps_queue_low¶

Since: 13.12.0, 14.1.0

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.

taskprocessor_overload_trigger¶

Since: 13.26.0, 16.3.0

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.

• global - (default) Any taskprocessor overload will trigger.
  

• pjsip_only - Only pjsip taskprocessor overloads will trigger.
  

• none - No overload detection will be performed.
  

unidentified_request_count¶

Since: 13.10.0

If 'unidentified_request_count' unidentified requests are received during 'unidentified_request_period', a security event will be generated.

unidentified_request_period¶

Since: 13.10.0

If 'unidentified_request_count' unidentified requests are received during 'unidentified_request_period', a security event will be generated.

use_callerid_contact¶

Since: 13.24.0, 16.1.0

This option will cause Asterisk to place caller-id information into generated Contact headers.

Generated Version¶

This documentation was generated from Asterisk branch 20 using version GIT