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Option Name	Type	Default Value	Regular Expression	Description
`100rel`	`Custom`	`yes`	`false`	Allow support for RFC3262 provisional ACK tags
`aggregate_mwi`	`Boolean`	`yes`	`false`	Condense MWI notifications into a single NOTIFY.
`allow`	`Codec`		`false`	Media Codec(s) to allow
`allow_overlap`	`Boolean`	`yes`	`false`	Enable RFC3578 overlap dialing support.
`aors`	`String`		`false`	AoR(s) to be used with the endpoint
`auth`	`Custom`		`false`	Authentication Object(s) associated with the endpoint
`callerid`	`Custom`		`false`	CallerID information for the endpoint
`callerid_privacy`	`Custom`	`allowed_not_screened`	`false`	Default privacy level
`callerid_tag`	`Custom`		`false`	Internal id_tag for the endpoint
`context`	`String`	`default`	`false`	Dialplan context for inbound sessions
`direct_media_glare_mitigation`	`Custom`	`none`	`false`	Mitigation of direct media (re)INVITE glare
`direct_media_method`	`Custom`	`invite`	`false`	Direct Media method type
`trust_connected_line`	`Boolean`	`yes`	`false`	Accept Connected Line updates from this endpoint
`send_connected_line`	`Boolean`	`yes`	`false`	Send Connected Line updates to this endpoint
`connected_line_method`	`Custom`	`invite`	`false`	Connected line method type
`direct_media`	`Boolean`	`yes`	`false`	Determines whether media may flow directly between endpoints.
`disable_direct_media_on_nat`	`Boolean`	`no`	`false`	Disable direct media session refreshes when NAT obstructs the media session
`disallow`				Media Codec(s) to disallow
`dtmf_mode`	`Custom`	`rfc4733`	`false`	DTMF mode
`media_address`	`String`		`false`	IP address used in SDP for media handling
`bind_rtp_to_media_address`	`Boolean`	`no`	`false`	Bind the RTP instance to the media_address
`force_rport`	`Boolean`	`yes`	`false`	Force use of return port
`ice_support`	`Boolean`	`no`	`false`	Enable the ICE mechanism to help traverse NAT
`identify_by`	`Custom`	`username,ip`	`false`	Way(s) for Endpoint the endpoint to be identified
`redirect_method`	`Custom`	`user`	`false`	How redirects received from an endpoint are handled
`mailboxes`	`String`		`false`	NOTIFY the endpoint when state changes for any of the specified mailboxes
`mwi_subscribe_replaces_unsolicited`	`Boolean`	`no`	`false`	An MWI subscribe will replace sending unsolicited NOTIFYs
`voicemail_extension`	`Custom`		`false`	The voicemail extension to send in the NOTIFY Message-Account header
`moh_suggest`	`String`	`default`	`false`	Default Music On Hold class
`outbound_auth`	`Custom`		`false`	Authentication object(s) used for outbound requests
`outbound_proxy`	`String`		`false`	Proxy through which Full SIP URI of the outbound proxy used to send requests , a full SIP URI must be provided
`rewrite_contact`	`Boolean`	`no`	`false`	Allow Contact header to be rewritten with the source IP address-port
`rtp_ipv6`	`Boolean`	`no`	`false`	Allow use of IPv6 for RTP traffic
`rtp_symmetric`	`Boolean`	`no`	`false`	Enforce that RTP must be symmetric
`send_diversion`	`Boolean`	`yes`	`false`	Send the Diversion header, conveying the diversion information to the called user agent
`send_history_info`	`Boolean`	`no`	`false`	Send the History-Info header, conveying the diversion information to the called and calling user agents
`send_pai`	`Boolean`	`no`	`false`	Send the P-Asserted-Identity header
`send_rpid`	`Boolean`	`no`	`false`	Send the Remote-Party-ID header
`rpid_immediate`	`Boolean`	`no`	`false`	Immediately send connected line updates on unanswered incoming calls.
`timers_min_se`	`Unsigned Integer`	`90`	`false`	Minimum session timers expiration period
`timers`	`Custom`	`yes`	`false`	Session timers for SIP packets
`timers_sess_expires`	`Unsigned Integer`	`1800`	`false`	Maximum session timer expiration period
`transport`	`String`		`false`	Desired Explicit transport configuration to use
`trust_id_inbound`	`Boolean`	`no`	`false`	Accept identification information received from this endpoint
`trust_id_outbound`	`Boolean`	`no`	`false`	Send private identification details to the endpoint.
`type`	`None`		`false`	Must be of type 'endpoint'.
`use_ptime`	`Boolean`	`no`	`false`	Use Endpoint's requested packetisation packetization interval
`use_avpf`	`Boolean`	`no`	`false`	Determines whether res_pjsip will use and enforce usage of AVPF for this endpoint.
`force_avp`	`Boolean`	`no`	`false`	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`	`Boolean`	`no`	`false`	Determines whether res_pjsip will use the media transport received in the offer SDP in the corresponding answer SDP.
`media_encryption`	`Custom`	`no`	`false`	Determines whether res_pjsip will use and enforce usage of media encryption for this endpoint.
`media_encryption_optimistic`	`Boolean`	`no`	`false`	Determines whether encryption should be used if possible but does not terminate the session if not achieved.
`g726_non_standard`	`Boolean`	`no`	`false`	Force g.726 to use AAL2 packing order when negotiating g.726 audio
`inband_progress`	`Boolean`	`no`	`false`	Determines whether chan_pjsip will indicate ringing using inband progress.
`call_group`	`Custom`		`false`	The numeric pickup groups for a channel.
`pickup_group`	`Custom`		`false`	The numeric pickup groups that a channel can pickup.
`named_call_group`	`Custom`		`false`	The named pickup groups for a channel.
`named_pickup_group`	`Custom`		`false`	The named pickup groups that a channel can pickup.
`device_state_busy_at`	`Unsigned Integer`	`0`	`false`	The number of in-use channels which will cause busy to be returned as device state
`t38_udptl`	`Boolean`	`no`	`false`	Whether T.38 UDPTL support is enabled or not
`t38_udptl_ec`	`Custom`	`none`	`false`	T.38 UDPTL error correction method
`t38_udptl_maxdatagram`	`Unsigned Integer`	`0`	`false`	T.38 UDPTL maximum datagram size
`fax_detect`	`Boolean`	`no`	`false`	Whether CNG tone detection is enabled
`fax_detect_timeout`	`Unsigned Integer`	`0`	`false`	How long into a call before fax_detect is disabled for the call
`t38_udptl_nat`	`Boolean`	`no`	`false`	Whether NAT support is enabled on UDPTL sessions
`t38_udptl_ipv6`	`Boolean`	`no`	`false`	Whether IPv6 is used for UDPTL Sessions
`tone_zone`	`String`		`false`	Set which country's indications to use for channels created for this endpoint.
`language`	`String`		`false`	Set the default language to use for channels created for this endpoint.
`one_touch_recording`	`Boolean`	`no`	`false`	Determines whether one-touch recording is allowed for this endpoint.
`record_on_feature`	`String`	`automixmon`	`false`	The feature to enact when one-touch recording is turned on.
`record_off_feature`	`String`	`automixmon`	`false`	The feature to enact when one-touch recording is turned off.
`rtp_engine`	`String`	`asterisk`	`false`	Name of the RTP engine to use for channels created for this endpoint
`allow_transfer`	`Boolean`	`yes`	`false`	Determines whether SIP REFER transfers are allowed for this endpoint
`user_eq_phone`	`Boolean`	`no`	`false`	Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number
`moh_passthrough`	`Boolean`	`no`	`false`	Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side
`sdp_owner`	`String`	`-`	`false`	String placed as the username portion of an SDP origin (o=) line.
`sdp_session`	`String`	`Asterisk`	`false`	String used for the SDP session (s=) line.
`tos_audio`	`Custom`	`0`	`false`	DSCP TOS bits for audio streams
`tos_video`	`Custom`	`0`	`false`	DSCP TOS bits for video streams
`cos_audio`	`Unsigned Integer`	`0`	`false`	Priority for audio streams
`cos_video`	`Unsigned Integer`	`0`	`false`	Priority for video streams
`allow_subscribe`	`Boolean`	`yes`	`false`	Determines if endpoint is allowed to initiate subscriptions with Asterisk.
`sub_min_expiry`	`Unsigned Integer`	`0`	`false`	The minimum allowed expiry time for subscriptions initiated by the endpoint.
`from_user`	`String` `Custom`		`false`	Username to use in From header for requests to this endpoint.
`mwi_from_user`	`String`		`false`	Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.
`from_domain`	`String`		`false`	Domain to user in From header for requests to this endpoint.
`dtls_verify`	`Custom`	`no`	`false`	Verify that the provided peer certificate is valid
`dtls_rekey`	`Custom`	`0`	`false`	Interval at which to renegotiate the TLS session and rekey the SRTP session
`dtls_cert_file`	`Custom`		`false`	Path to certificate file to present to peer
`dtls_private_key`	`Custom`		`false`	Path to private key for certificate file
`dtls_cipher`	`Custom`		`false`	Cipher to use for DTLS negotiation
`dtls_ca_file`	`Custom`		`false`	Path to certificate authority certificate
`dtls_ca_path`	`Custom`		`false`	Path to a directory containing certificate authority certificates
`dtls_setup`	`Custom`		`false`	Whether we are willing to accept connections, connect to the other party, or both.
`dtls_fingerprint`	`Custom`		`false`	Type of hash to use for the DTLS fingerprint in the SDP.
`srtp_tag_32`	`Boolean`	`no`	`false`	Determines whether 32 byte tags should be used instead of 80 byte tags.
`set_var`	`Custom`		`false`	Variable set on a channel involving the endpoint.
`message_context`	`String`		`false`	Context to route incoming MESSAGE requests to.
`accountcode`	`String`		`false`	An accountcode to set automatically on any channels created for this endpoint.
`rtp_keepalive`	`Unsigned Integer`	`0`	`false`	Number of seconds between RTP comfort noise keepalive packets.
`rtp_timeout`	`Unsigned Integer`	`0`	`false`	Maximum number of seconds without receiving RTP (while off hold) before terminating call.
`rtp_timeout_hold`	`Unsigned Integer`	`0`	`false`	Maximum number of seconds without receiving RTP (while on hold) before terminating call.
`acl`	`Custom`		`false`	List of IP ACL section names in acl.conf
`deny`	`Custom`		`false`	List of IP addresses to deny access from
`permit`	`Custom`		`false`	List of IP addresses to permit access from
`contact_acl`	`Custom`		`false`	List of Contact ACL section names in acl.conf
`contact_deny`	`Custom`		`false`	List of Contact header addresses to deny
`contact_permit`	`Custom`		`false`	List of Contact header addresses to permit
`subscribe_context`	`String`		`false`	Context for incoming MESSAGE requests.
`contact_user`	`Custom`		`false`	Force the user on the outgoing Contact header to this value.
`asymmetric_rtp_codec`	`Boolean`	`no`	`false`	Allow the sending and receiving RTP codec to differ
`rtcp_mux`	`Boolean`	`no`	`false`	Enable RFC 5761 RTCP multiplexing on the RTP port
`refer_blind_progress`	`Boolean`	`yes`	`false`	Whether to notifies all the progress details on blind transfer
`notify_early_inuse_ringing`	`Boolean`	`no`	`false`	Whether to notifies dialog-info 'early' on InUse&Ringing state
`incoming_mwi_mailbox`	`String`		`false`	Mailbox name to use when incoming MWI NOTIFYs are received
`follow_early_media_fork`	`Boolean`	`yes`	`false`	Follow SDP forked media when To tag is different
`accept_multiple_sdp_answers`	`Boolean`	`no`	`false`	Accept multiple SDP answers on non-100rel responses
`suppress_q850_reason_headers`	`Boolean`	`no`	`false`	Suppress Q.850 Reason headers for this endpoint
`ignore_183_without_sdp`	`Boolean`	`no`	`false`	Do not forward 183 when it doesn't contain SDP

Configuration Option Descriptions

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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.verification.

Info

title	Note

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.

Anchor

	endpoint_callerid
	endpoint_callerid

...

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.

Anchor

	endpoint_dtmf_mode
	endpoint_dtmf_mode

...

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.

Anchor

	endpoint_media_address
	endpoint_media_address

...

Info

title	Note

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

Anchor

	endpoint_bind_rtp_to_media_address
	endpoint_bind_rtp_to_media_address

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.

Anchor

	endpoint_identify_by
	endpoint_identify_by

identify_by

An endpoint Endpoints and AORs can be identified in multiple ways. Currently, the only supported option is username, which matches the endpoint This option is a comma separated list of methods the endpoint can be identified.

Info

title	Note

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.

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.
Info
title 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

...

username

for more details on this method of endpoint identification.

Anchor

	endpoint_redirect_method
	endpoint_redirect_method

...

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_mwi_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.

Anchor

	endpoint_outbound_auth
	endpoint_outbound_auth

outbound_auth

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

Info

title	Note

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.

Anchor

	endpoint_rewrite_contact
	endpoint_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 send 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.

Anchor

	endpoint_rpid_immediate
	endpoint_rpid_immediate

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.

Anchor

	endpoint_timers_min_se
	endpoint_timers_min_se

timers_min_se

Minimium Minimum session timer expiration period. Time in seconds.

Anchor

	endpoint_timers
	endpoint_timers

timers

forced
no
yes
required
yes
always
forced - Alias of always

Anchor

	endpoint_timers_sess_expires
	endpoint_timers_sess_expires

timers_sess_expires

Maximium Maximum session timer expiration period. Time in seconds.

Anchor

	endpoint_transport
	endpoint_transport

transport

This will set the desired force the endpoint to use the specified transport configuration to send SIP data through.

...

messages. You need to already know what kind of transport (UDP/TCP/IPv4/etc) the endpoint device will use.

Info

title	WarningNote

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

...

If set to no, res_pjsip will use the AVP or SAVP RTP profile for all media offers on outbound calls and media updates, but will accept either and will decline media offers not using the AVP /AVPF or SAVP /SAVPF RTP profile for all inbound media offers.

Anchor

	endpoint_force_avp
	endpoint_force_avp

...

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.

Anchor

	endpoint_media_encryption_optimistic
	endpoint_media_encryption_optimistic

media_encryption_optimistic

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

Anchor

	endpoint_g726_non_standard
	endpoint_g726_non_standard

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.

Anchor

	endpoint_inband_progress
	endpoint_inband_progress

...

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

...

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

Anchor

	endpoint_fax_detect_timeout
	endpoint_fax_detect_timeout

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.

Anchor

	endpoint_t38_udptl_nat
	endpoint_t38_udptl_nat

...

This option only applies if media_encryption is set to dtls.

It can be one of the following values:

no - meaning no verificaton 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.

Anchor

	endpoint_dtls_rekey
	endpoint_dtls_rekey

...

Many options for acceptable ciphers. See link for more:

http://www.openssl.org/docs/apps/ciphers.html#CIPHER\_STRINGS

...

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.

Anchor

	endpoint_dtls_fingerprint
	endpoint_dtls_fingerprint

dtls_fingerprint

This option only applies if media_encryption is set to dtls.

SHA-256
SHA-1

Anchor

	endpoint_srtp_tag_32
	endpoint_srtp_tag_32

...

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

Anchor

	endpoint_rtp_keepalive
	endpoint_rtp_keepalive

rtp_keepalive

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.

Anchor

	endpoint_rtp_timeout
	endpoint_rtp_timeout

rtp_timeout

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.

Anchor

	endpoint_rtp_timeout_hold
	endpoint_rtp_timeout_hold

rtp_timeout_hold

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.

Anchor

	endpoint_acl
	endpoint_acl

acl

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

Anchor

	endpoint_deny
	endpoint_deny

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 ('/')

Anchor

	endpoint_permit
	endpoint_permit

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 ('/')

Anchor

	endpoint_contact_acl
	endpoint_contact_acl

contact_acl

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

Anchor

	endpoint_contact_deny
	endpoint_contact_deny

contact_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 ('/')

Anchor

	endpoint_contact_permit
	endpoint_contact_permit

contact_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 ('/')

Anchor

	endpoint_subscribe_context
	endpoint_subscribe_context

subscribe_context

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.

Anchor

	endpoint_contact_user
	endpoint_contact_user

contact_user

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

Anchor

	endpoint_asymmetric_rtp_codec
	endpoint_asymmetric_rtp_codec

asymmetric_rtp_codec

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.

Anchor

	endpoint_rtcp_mux
	endpoint_rtcp_mux

rtcp_mux

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.

Anchor

	endpoint_refer_blind_progress
	endpoint_refer_blind_progress

refer_blind_progress

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".

Anchor

	endpoint_notify_early_inuse_ringing
	endpoint_notify_early_inuse_ringing

notify_early_inuse_ringing

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

Anchor

	endpoint_incoming_mwi_mailbox
	endpoint_incoming_mwi_mailbox

incoming_mwi_mailbox

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.

Anchor

	endpoint_follow_early_media_fork
	endpoint_follow_early_media_fork

follow_early_media_fork

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.

Info

title	Note

This option must also be enabled in the system section for it to take effect here.

Anchor

	endpoint_accept_multiple_sdp_answers
	endpoint_accept_multiple_sdp_answers

accept_multiple_sdp_answers

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.

Info

title	Note

This option must also be enabled in the system section for it to take effect here.

Anchor

	endpoint_suppress_q850_reason_headers
	endpoint_suppress_q850_reason_headers

suppress_q850_reason_headers

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.

Anchor

	endpoint_ignore_183_without_sdp
	endpoint_ignore_183_without_sdp

ignore_183_without_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.

auth

Authentication type

Configuration Option Reference

Option Name	Type	Default Value	Regular Expression	Description
`auth_type`	`Custom`	`userpass`	`false`	Authentication type
`nonce_lifetime`	`Unsigned Integer`	`32`	`false`	Lifetime of a nonce associated with this authentication config.
`md5_cred`	`String`		`false`	MD5 Hash used for authentication.
`password`	`String`		`false`	PlainText Plain text password used for authentication.
`realm`	`String`		`false`	SIP realm for endpoint
`type`	`None`		`false`	Must be 'auth'
`username`	`String`		`false`	Username to use for account

...

Only used when auth_type is userpass.

Anchor

	auth_realm
	auth_realm

realm

The treatment of this value depends upon how the authentication object is used.

When used as an inbound authentication object, the realm is sent as part of the challenge so the peer can know which key to use when responding. An empty value will use the global section's default_realm value when issuing a challenge.

When used as an outbound authentication object, the realm is matched with the received challenge realm to determine which authentication object to use when responding to the challenge. An empty value matches any challenging realm when determining which authentication object matches a received challenge.

Info

title	Note

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.

domain_alias

Domain Alias

Configuration Option Reference

...

Option Name	Type	Default Value	Regular Expression	Description
`async_operations`	`Unsigned Integer`	`1`	`false`	Number of simultaneous Asynchronous Operations
`bind`	`Custom`		`false`	IP Address and optional port to bind to for this transport
`ca_list_file`	`String` `Custom`		`false`	File containing a list of certificates to read (TLS ONLY, not WSS)
`ca_list_path`	`Custom`		`false`	Path to directory containing a list of certificates to read (TLS ONLY, not WSS)
`cert_file`	`String` `Custom`		`false`	Certificate file for endpoint (TLS ONLY, not WSS)
`cipher`	`Custom`		`false`	Preferred Cryptography Cipher cryptography cipher names (TLS ONLY, not WSS)
`domain`	`String`		`false`	Domain the transport comes from
`external_media_address`	`String`		`false`	External IP address to use in RTP handling
`external_signaling_address`	`String`		`false`	External address for SIP signalling
`external_signaling_port`	`Unsigned Integer`	`0`	`false`	External port for SIP signalling
`method`	`Custom`		`false`	Method of SSL transport (TLS ONLY, not WSS)
`local_net`	`Custom`		`false`	Network to consider local (used for NAT purposes).
`password`	`String`		`false`	Password required for transport
`priv_key_file`	`String` `Custom`		`false`	Private key file (TLS ONLY, not WSS)
`protocol`	`Custom`	`udp`	`false`	Protocol to use for SIP traffic
`require_client_cert`	`Custom`		`false`	Require client certificate (TLS ONLY, not WSS)
`type`	`None` `Custom`		`false`	Must be of type 'transport'.
`verify_client`	`Custom`		`false`	Require verification of client certificate (TLS ONLY, not WSS)
`verify_server`	`Custom`		`false`	Require verification of server certificate (TLS ONLY, not WSS)
`tos`	`Custom`	`0`	`false`	Enable TOS for the signalling sent over this transport
`cos`	`Unsigned Integer`	`0`	`false`	Enable COS for the signalling sent over this transport
`websocket_write_timeout`	`Integer`	`100`	`false`	The timeout (in milliseconds) to set on WebSocket connections.
`allow_reload`	`Boolean`	`no`	`false`	Allow this transport to be reloaded.
`symmetric_transport`	`Boolean`	`no`	`false`	Use the same transport for outgoing requests as incoming ones.

Configuration Option Descriptions

Anchor

	transport_cert_file
	transport_cert_file

cert_file

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.

Anchor

	transport_cipher
	transport_cipher

cipher

Many options for acceptable ciphers see 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\_STRINGSSUITE\_NAMES

Anchor

	transport_external_media_address
	transport_external_media_address

...

Anchor

	transport_method
	transport_method

method

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
sslv2
sslv3
sslv23

Anchor

	transport_local_net
	transport_local_net

...

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; default is 100 ms.

Anchor

	transport_allow_reload
	transport_allow_reload

allow_reload

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.

Anchor

	transport_symmetric_transport
	transport_symmetric_transport

symmetric_transport

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.

contact

A way of creating an aliased name to a SIP URI

...

Option Name	Type	Default Value	Regular Expression	Description
`type`	`None`		`false`	Must be of type 'contact'.
`uri`	`String`		`false`	SIP URI to contact peer
`expiration_time`	`Custom`		`false`	Time to keep alive a contact
`qualify_frequency`	`Unsigned Integer`	`0`	`false`	Interval at which to qualify a contact
`qualify_timeout`	`Double`	`3.0`	`false`	Timeout for qualify
`authenticate_qualify`	`Boolean`	`no`	`false`	Authenticates a qualify challenge response if needed
`outbound_proxy`	`String`		`false`	Outbound proxy used when sending OPTIONS request
`path`	`String`		`false`	Stored Path vector for use in Route headers on outgoing requests.
`user_agent`	`String`		`false`	User-Agent header from registration.
`endpoint`	`String`		`false`	Endpoint name
`reg_server`	`String`		`false`	Asterisk Server name
`via_addr`	`String`		`false`	IP-address of the last Via header from registration.
`via_port`	`Unsigned Integer`	`0`	`false`	IP-port of the last Via header from registration.
`call_id`	`String`		`false`	Call-ID header from registration.
`prune_on_boot`	`Boolean`	`no`	`false`	A contact that cannot survive a restart/boot.

Configuration Option Descriptions

...

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

Anchor

	contact_qualify_timeout
	contact_qualify_timeout

qualify_timeout

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.

Anchor

	contact_authenticate_qualify
	contact_authenticate_qualify

authenticate_qualify

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

Info

title	Note

This option does nothing as we will always complete the challenge response authentication if the qualify request is challenged.

Anchor

	contact_outbound_proxy
	contact_outbound_proxy

...

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

Anchor

	contact_endpoint
	contact_endpoint

endpoint

The name of the endpoint this contact belongs to

Anchor

	contact_reg_server
	contact_reg_server

reg_server

Asterisk Server name on which SIP endpoint registered.

Anchor

	contact_via_addr
	contact_via_addr

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.

Anchor

	contact_via_port
	contact_via_port

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.

Anchor

	contact_call_id
	contact_call_id

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.

Anchor

	contact_prune_on_boot
	contact_prune_on_boot

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.

aor

The configuration for a location of an endpoint

...

Option Name	Type	Default Value	Regular Expression	Description
`contact`	`Custom`		`false`	Permanent contacts assigned to AoR
`default_expiration`	`Unsigned Integer`	`3600`	`false`	Default expiration time in seconds for contacts that are dynamically bound to an AoR.
`mailboxes`	`String`		`false`	Allow subscriptions for the specified mailbox(es)
`voicemail_extension`	`Custom`		`false`	The voicemail extension to send in the NOTIFY Message-Account header
`maximum_expiration`	`Unsigned Integer`	`7200`	`false`	Maximum time to keep an AoR
`max_contacts`	`Unsigned Integer`	`0`	`false`	Maximum number of contacts that can bind to an AoR
`minimum_expiration`	`Unsigned Integer`	`60`	`false`	Minimum keep alive time for an AoR
`remove_existing`	`Boolean`	`no`	`false`	Determines whether new contacts replace existing ones.
`type`	`None`		`false`	Must be of type 'aor'.
`qualify_frequency`	`Unsigned Integer`	`0`	`false`	Interval at which to qualify an AoR
`qualify_timeout`	`Double`	`3.0`	`false`	Timeout for qualify
`authenticate_qualify`	`Boolean`	`no`	`false`	Authenticates a qualify request challenge response if needed
`outbound_proxy`	`String`		`false`	Outbound proxy used when sending OPTIONS request
`support_path`	`Boolean`	`no`	`false`	Enables Path support for REGISTER requests and Route support for other requests.

...

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

...

Anchor

	aor_maximum_expiration
	aor_maximum_expiration

maximum_expiration

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

...

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.

Info

title	Note

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.

Info

title	Note

This should be set to 1 and remove_existing set to yes if you wish to stick with the older chan_sip behaviour.

...

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

...

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.

Info

title	Note

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.

Info

title	Note

This should be set to yes and max_contacts set to 1 if you wish to stick with the older chan_sip behaviour.

...

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

Anchor

	aor_qualify_timeout
	aor_qualify_timeout

qualify_timeout

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.

Anchor

	aor_authenticate_qualify
	aor_authenticate_qualify

authenticate_qualify

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

Info

title	Note

This option does nothing as we will always complete the challenge response authentication if the qualify request is challenged.

Anchor

	aor_outbound_proxy
	aor_outbound_proxy

...

Option Name	Type	Default Value	Regular Expression	Description
`timer_t1`	`Unsigned Integer`	`500`	`false`	Set transaction timer T1 value (milliseconds).
`timer_b`	`Unsigned Integer`	`32000`	`false`	Set transaction timer B value (milliseconds).
`compact_headers`	`Boolean`	`no`	`false`	Use the short forms of common SIP header names.
`threadpool_initial_size`	`Unsigned Integer`	`0`	`false`	Initial number of threads in the res_pjsip threadpool.
`threadpool_auto_increment`	`Unsigned Integer`	`5`	`false`	The amount by which the number of threads is incremented when necessary.
`threadpool_idle_timeout`	`Unsigned Integer`	`60`	`false`	Number of seconds before an idle thread should be disposed of.
`threadpool_max_size`	`Unsigned Integer`	`0` `50`	`false`	Maximum number of threads in the res_pjsip threadpool. A value of 0 indicates no maximum.
`disable_tcp_switch`	`Boolean`	`yes`	`false`	Disable automatic switching from UDP to TCP transports.
`follow_early_media_fork`	`Boolean`	`yes`	`false`	Follow SDP forked media when To tag is different
`accept_multiple_sdp_answers`	`Boolean`	`no`	`false`	Follow SDP forked media when To tag is the same
`disable_rport`	`Boolean`	`no`	`false`	Disable the use of rport in outgoing requests.
`type`	`None`		`false`	Must be of type 'system' UNLESS the object name is 'system'.

Configuration Option Descriptions

...

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.

Anchor

	system_disable_tcp_switch
	system_disable_tcp_switch

disable_tcp_switch

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

Anchor

	system_follow_early_media_fork
	system_follow_early_media_fork

follow_early_media_fork

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.

Info

title	Note

This option must also be enabled on endpoints that require this functionality.

Anchor

	system_accept_multiple_sdp_answers
	system_accept_multiple_sdp_answers

accept_multiple_sdp_answers

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.

Info

title	Note

This option must also be enabled on endpoints that require this functionality.

Anchor

	system_disable_rport
	system_disable_rport

disable_rport

Remove "rport" parameter from the outgoing requests.

global

Options that apply globally to all SIP communications

...

Option Name	Type	Default Value	Regular Expression	Description
`max_forwards`	`Unsigned Integer`	`70`	`false`	Value used in Max-Forwards header for SIP requests.
`keep_alive_interval`	`Unsigned Integer`	`90`	`false`	The interval (in seconds) to send keepalives to active connection-oriented transports.
`contact_expiration_check_interval`	`Unsigned Integer`	`30`	`false`	The interval (in seconds) to check for expired contacts.
`disable_multi_domain`	`Boolean`	`no`	`false`	Disable Multi Domain support
`max_initial_qualify_time`	`Unsigned Integer`	`0`	`false`	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.
`unidentified_request_period`	`Unsigned Integer`	`5`	`false`	The number of seconds over which to accumulate unidentified requests.
`unidentified_request_count`	`Unsigned Integer`	`5`	`false`	The number of unidentified requests from a single IP to allow.
`unidentified_request_prune_interval`	`Unsigned Integer`	`30`	`false`	The interval at which unidentified requests are older than twice the unidentified_request_period are pruned.
`type`	`None`		`false`	Must be of type 'global' UNLESS the object name is 'global'.
`user_agent`	`String`	`Asterisk PBX SVNGIT-branch13-13-r423723.15.0-rc1-2825-g408d08a`	`false`	Value used in User-Agent header for SIP requests and Server header for SIP responses.
`regcontext`	`String`		`false`	When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us.
`default_outbound_endpoint`	`String`	`default_outbound_endpoint`	`false`	Endpoint to use when sending an outbound request to a URI without a specified endpoint.
`default_voicemail_extension`	`String`		`false`	The voicemail extension to send in the NOTIFY Message-Account header if not specified on endpoint or aor
`debug`	`String`	`no`	`false`	Enable/Disable SIP debug logging. Valid options include yes, no, or a host address
`endpoint_identifier_order`	`String`	`ip,username,anonymous`	`false`	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.
`default_from_user`	`String`	`asterisk`	`false`	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.
`default_realm`	`String`	`asterisk`	`false`	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_tps_queue_high`	`Unsigned Integer`	`500`	`false`	MWI taskprocessor high water alert trigger level.
`mwi_tps_queue_low`	`Integer`	`-1`	`false`	MWI taskprocessor low water clear alert level.
`mwi_disable_initial_unsolicited`	`Boolean`	`no`	`false`	Enable/Disable sending unsolicited MWI to all endpoints on startup.
`ignore_uri_user_options`	`Boolean`	`no`	`false`	Enable/Disable ignoring SIP URI user field options.
`use_callerid_contact`	`Boolean`	`no`	`false`	Place caller-id information into Contact header
`send_contact_status_on_update_registration`	`Boolean`	`yes`	`false`	Enable sending AMI ContactStatus event when a device refreshes its registration.
`taskprocessor_overload_trigger`	`Custom`	`global`	`false`	Trigger scope for taskprocessor overloads
`norefersub`	`Boolean`	`yes`	`false`	Advertise support for RFC4488 REFER subscription suppression

Configuration Option Descriptions

Anchor

	global_disable_multi_domain
	global_disable_multi_domain

disable_multi_domain

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

Anchor

	global_unidentified_request_period
	global_unidentified_request_period

unidentified_request_period

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

Anchor

	global_unidentified_request_count
	global_unidentified_request_count

unidentified_request_count

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

Anchor

	global_endpoint_identifier_order
	global_endpoint_identifier_order

endpoint_identifier_order

Info

title	Note

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.

Anchor

	global_mwi_tps_queue_high
	global_mwi_tps_queue_high

mwi_tps_queue_high

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.

Anchor

	global_mwi_tps_queue_low
	global_mwi_tps_queue_low

mwi_tps_queue_low

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.

Info

title	Note

Set to -1 for the low water level to be 90% of the high water level.

Anchor

	global_mwi_disable_initial_unsolicited
	global_mwi_disable_initial_unsolicited

mwi_disable_initial_unsolicited

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.

Anchor

	global_ignore_uri_user_options
	global_ignore_uri_user_options

ignore_uri_user_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.

Code Block

title	Example: Sample SIP URI
linenumbers	true



sip:1235557890;[email protected];user=phone

Code Block

title	Example: Sample SIP URI user field
linenumbers	true



1235557890;phone-context=national

Code Block

title	Example: Sample SIP URI user field truncated
linenumbers	true

1235557890

Info

title	Note

The caller-id and redirecting number strings obtained from incoming SIP URI user fields are always truncated at the first semicolon.

Anchor

	global_use_callerid_contact
	global_use_callerid_contact

use_callerid_contact

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

Anchor

	global_taskprocessor_overload_trigger
	global_taskprocessor_overload_trigger

taskprocessor_overload_trigger

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.
Warning
title Warning
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.

Import Version

This documentation was imported from Asterisk Version SVNGIT-branch13-13-r423723.15.0-rc1-2825-g408d08a

Page History

Versions Compared

Old Version 29

New Version Current

Key

Configuration Option Descriptions

bind_rtp_to_media_address

identify_by

outbound_auth

rpid_immediate

timers_min_se

timers

timers_sess_expires

transport

media_encryption_optimistic

g726_non_standard

fax_detect_timeout

dtls_fingerprint

rtp_keepalive

rtp_timeout

rtp_timeout_hold

acl

deny

permit

contact_acl

contact_deny

contact_permit

subscribe_context

contact_user

asymmetric_rtp_codec

rtcp_mux

refer_blind_progress

notify_early_inuse_ringing

incoming_mwi_mailbox

follow_early_media_fork

accept_multiple_sdp_answers

suppress_q850_reason_headers

ignore_183_without_sdp

auth

Configuration Option Reference

realm

domain_alias

Configuration Option Reference

Configuration Option Descriptions

cert_file

cipher

method

allow_reload

symmetric_transport

contact

Configuration Option Descriptions

qualify_timeout

authenticate_qualify

endpoint

reg_server

via_addr

via_port

call_id

prune_on_boot

aor

maximum_expiration

qualify_timeout

authenticate_qualify

Configuration Option Descriptions

disable_tcp_switch

follow_early_media_fork

accept_multiple_sdp_answers

disable_rport

global

Configuration Option Descriptions

disable_multi_domain

unidentified_request_period

unidentified_request_count

endpoint_identifier_order

mwi_tps_queue_high

mwi_tps_queue_low

mwi_disable_initial_unsolicited

ignore_uri_user_options

use_callerid_contact

taskprocessor_overload_trigger