SIP Resource using PJProject

This configuration documentation is for functionality provided by res_pjsip.

pjsip.conf

endpoint

Endpoint

Configuration Option Reference

Option Name	Default Value	Description
`100rel`	`yes`	Allow support for RFC3262 provisional ACK tags
`aggregate_mwi`	`yes`	Condense MWI notifications into a single NOTIFY.
`allow`		Media Codec(s) to allow
`aors`		AoR(s) to be used with the endpoint
`auth`		Authentication Object(s) associated with the endpoint
`callerid`		CallerID information for the endpoint
`callerid_privacy`		Default privacy level
`callerid_tag`		Internal id_tag for the endpoint
`context`		Dialplan context for inbound sessions
`direct_media_glare_mitigation`	`none`	Mitigation of direct media (re)INVITE glare
`direct_media_method`	`invite`	Direct Media method type
`connected_line_method`	`invite`	Connected line method type
`direct_media`	`yes`	Determines whether media may flow directly between endpoints.
`disable_direct_media_on_nat`	`no`	Disable direct media session refreshes when NAT obstructs the media session
`disallow`		Media Codec(s) to disallow
`dtmf_mode`	`rfc4733`	DTMF mode
`media_address`		IP address used in SDP for media handling
`bind_rtp_to_media_address`		Bind the RTP instance to the media_address
`force_rport`	`yes`	Force use of return port
`ice_support`	`no`	Enable the ICE mechanism to help traverse NAT
`identify_by`	`username,location`	Way(s) for Endpoint to be identified
`redirect_method`		How redirects received from an endpoint are handled
`mailboxes`		NOTIFY the endpoint when state changes for any of the specified mailboxes
`mwi_subscribe_replaces_unsolicited`		An MWI subscribe will replace sending unsolicited NOTIFYs
`voicemail_extension`		The voicemail extension to send in the NOTIFY Message-Account header
`moh_suggest`	`default`	Default Music On Hold class
`outbound_auth`		Authentication object used for outbound requests
`outbound_proxy`		Proxy through which to send requests, a full SIP URI must be provided
`rewrite_contact`		Allow Contact header to be rewritten with the source IP address-port
`rtp_ipv6`	`no`	Allow use of IPv6 for RTP traffic
`rtp_symmetric`	`no`	Enforce that RTP must be symmetric
`send_diversion`	`yes`	Send the Diversion header, conveying the diversion information to the called user agent
`send_pai`	`no`	Send the P-Asserted-Identity header
`send_rpid`	`no`	Send the Remote-Party-ID header
`rpid_immediate`	`no`	Immediately send connected line updates on unanswered incoming calls.
`timers_min_se`	`90`	Minimum session timers expiration period
`timers`	`yes`	Session timers for SIP packets
`timers_sess_expires`	`1800`	Maximum session timer expiration period
`transport`		Desired transport configuration
`trust_id_inbound`	`no`	Accept identification information received from this endpoint
`trust_id_outbound`	`no`	Send private identification details to the endpoint.
`type`		Must be of type 'endpoint'.
`use_ptime`	`no`	Use Endpoint's requested packetisation interval
`use_avpf`	`no`	Determines whether res_pjsip will use and enforce usage of AVPF for this endpoint.
`force_avp`	`no`	Determines whether res_pjsip will use and enforce usage of AVP, regardless of the RTP profile in use for this endpoint.
`media_use_received_transport`	`no`	Determines whether res_pjsip will use the media transport received in the offer SDP in the corresponding answer SDP.
`media_encryption`	`no`	Determines whether res_pjsip will use and enforce usage of media encryption for this endpoint.
`media_encryption_optimistic`	`no`	Determines whether encryption should be used if possible but does not terminate the session if not achieved.
`g726_non_standard`	`no`	Force g.726 to use AAL2 packing order when negotiating g.726 audio
`inband_progress`	`no`	Determines whether chan_pjsip will indicate ringing using inband progress.
`call_group`		The numeric pickup groups for a channel.
`pickup_group`		The numeric pickup groups that a channel can pickup.
`named_call_group`		The named pickup groups for a channel.
`named_pickup_group`		The named pickup groups that a channel can pickup.
`device_state_busy_at`	`0`	The number of in-use channels which will cause busy to be returned as device state
`t38_udptl`	`no`	Whether T.38 UDPTL support is enabled or not
`t38_udptl_ec`	`none`	T.38 UDPTL error correction method
`t38_udptl_maxdatagram`	`0`	T.38 UDPTL maximum datagram size
`fax_detect`	`no`	Whether CNG tone detection is enabled
`fax_detect_timeout`		How long into a call before fax_detect is disabled for the call
`t38_udptl_nat`	`no`	Whether NAT support is enabled on UDPTL sessions
`t38_udptl_ipv6`	`no`	Whether IPv6 is used for UDPTL Sessions
`tone_zone`		Set which country's indications to use for channels created for this endpoint.
`language`		Set the default language to use for channels created for this endpoint.
`one_touch_recording`	`no`	Determines whether one-touch recording is allowed for this endpoint.
`record_on_feature`	`automixmon`	The feature to enact when one-touch recording is turned on.
`record_off_feature`	`automixmon`	The feature to enact when one-touch recording is turned off.
`rtp_engine`	`asterisk`	Name of the RTP engine to use for channels created for this endpoint
`allow_transfer`	`yes`	Determines whether SIP REFER transfers are allowed for this endpoint
`user_eq_phone`	`no`	Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number
`sdp_owner`	`-`	String placed as the username portion of an SDP origin (o=) line.
`sdp_session`	`Asterisk`	String used for the SDP session (s=) line.
`tos_audio`		DSCP TOS bits for audio streams
`tos_video`		DSCP TOS bits for video streams
`cos_audio`		Priority for audio streams
`cos_video`		Priority for video streams
`allow_subscribe`	`yes`	Determines if endpoint is allowed to initiate subscriptions with Asterisk.
`sub_min_expiry`	`60`	The minimum allowed expiry time for subscriptions initiated by the endpoint.
`from_user`		Username to use in From header for requests to this endpoint.
`mwi_from_user`		Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.
`from_domain`		Domain to user in From header for requests to this endpoint.
`dtls_verify`		Verify that the provided peer certificate is valid
`dtls_rekey`		Interval at which to renegotiate the TLS session and rekey the SRTP session
`dtls_cert_file`		Path to certificate file to present to peer
`dtls_private_key`		Path to private key for certificate file
`dtls_cipher`		Cipher to use for DTLS negotiation
`dtls_ca_file`		Path to certificate authority certificate
`dtls_ca_path`		Path to a directory containing certificate authority certificates
`dtls_setup`		Whether we are willing to accept connections, connect to the other party, or both.
`dtls_fingerprint`		Type of hash to use for the DTLS fingerprint in the SDP.
`srtp_tag_32`		Determines whether 32 byte tags should be used instead of 80 byte tags.
`set_var`		Variable set on a channel involving the endpoint.
`message_context`		Context to route incoming MESSAGE requests to.
`accountcode`		An accountcode to set automatically on any channels created for this endpoint.
`rtp_keepalive`		Number of seconds between RTP comfort noise keepalive packets.
`rtp_timeout`	`0`	Maximum number of seconds without receiving RTP (while off hold) before terminating call.
`rtp_timeout_hold`	`0`	Maximum number of seconds without receiving RTP (while on hold) before terminating call.
`acl`		List of IP ACL section names in acl.conf
`deny`		List of IP addresses to deny access from
`permit`		List of IP addresses to permit access from
`contact_acl`		List of Contact ACL section names in acl.conf
`contact_deny`		List of Contact header addresses to deny
`contact_permit`		List of Contact header addresses to permit
`subscribe_context`		Context for incoming MESSAGE requests.
`contact_user`		Force the user on the outgoing Contact header to this value.
`asymmetric_rtp_codec`	`no`	Allow the sending and receiving RTP codec to differ

Configuration Option Descriptions

100rel

no
required
yes

aggregate_mwi

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.

aors

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

auth

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 vertification.

callerid

Must be in the format Name <Number>, or only <Number>.

callerid_privacy

allowed_not_screened
allowed_passed_screen
allowed_failed_screen
allowed
prohib_not_screened
prohib_passed_screen
prohib_failed_screen
prohib
unavailable

direct_media_glare_mitigation

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

none
outgoing
incoming

direct_media_method

Method for setting up Direct Media between endpoints.

invite
reinvite - Alias for the invite value.
update

connected_line_method

Method used when updating connected line information.

invite
reinvite - Alias for the invite value.
update

dtmf_mode

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.

media_address

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

Note

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Be aware that the external_media_address option, set in Transport configuration, can also affect the final media address used in the SDP.

bind_rtp_to_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.

identify_by

Endpoints and aors can be identified in multiple ways. Currently, the supported options are username, which matches the endpoint or aor id based on the username and domain in the From header (or To header for aors), and auth_username, which matches the endpoint or aor id based on the username and realm in the Authentication header. In all cases, if an exact match on both username and domain/realm fails, the match will be retried with just the username.

Note

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Identification by auth_username 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 in the "global" configuration object.

Note

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Endpoints can also be identified by IP address; however, that method of identification is not handled by this configuration option. See the documentation for the identify configuration section for more details on that method of endpoint identification. If this option is set and an identify configuration section exists for the endpoint, then the endpoint can be identified in multiple ways.

username
auth_username

redirect_method

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

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 [email protected]; for example: [email protected] For mailboxes provided by external sources, such as through the res_external_mwi module, you must specify strings supported by the external system.

For endpoints that SUBSCRIBE for MWI, use the mailboxes option in your AOR configuration.

rewrite_contact

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.

rpid_immediate

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.

timers_min_se

Minimium session timer expiration period. Time in seconds.

timers

no
yes
required
always
forced - Alias of always

timers_sess_expires

Maximium session timer expiration period. Time in seconds.

transport

This will set the desired transport configuration to send SIP data through.

Warning

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Not specifying a transport will DEFAULT to the first configured transport in pjsip.conf which is valid for the URI we are trying to contact.

Warning

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Transport configuration is not affected by reloads. In order to change transports, a full Asterisk restart is required

trust_id_inbound

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

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

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.

force_avp

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.

media_use_received_transport

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.

media_encryption

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

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

g726_non_standard

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.

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.

call_group

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

pickup_group

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

named_call_group

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

named_pickup_group

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

device_state_busy_at

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.

t38_udptl

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

t38_udptl_ec

none - No error correction should be used.
fec - Forward error correction should be used.
redundancy - Redundacy error correction should be used.

t38_udptl_maxdatagram

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

fax_detect

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

fax_detect_timeout

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.

t38_udptl_nat

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

t38_udptl_ipv6

When enabled the UDPTL stack will use IPv6.

record_on_feature

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.

Note

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This setting has no effect if the endpoint's one_touch_recording option is disabled

record_off_feature

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.

Note

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This setting has no effect if the endpoint's one_touch_recording option is disabled