# Asterisk Project : Asterisk 12 Configuration_res_pjsip

Added by wikibot , edited by wikibot on Aug 29, 2013

# SIP Resource using PJProject

This configuration documentation is for functionality provided by `res_pjsip`.

## pjsip.conf

### endpoint

Endpoint

#### Configuration Option Reference

Option Name | Type | Default Value | Regular Expression | Description
--- | --- | --- | --- | ---
`100rel` | `Custom` | `yes` | `false` | Allow support for RFC3262 provisional ACK tags
`aggregate_mwi` | `Boolean` | `yes` | `false` |
`allow` | `Codec` |  | `false` | Media Codec(s) to allow
`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` |  | `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
`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` | `Codec` |  | `false` | Media Codec(s) to disallow
`dtmfmode` | `Custom` | `rfc4733` | `false` | DTMF mode
`external_media_address` | `String` |  | `false` | IP used for External Media handling
`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` | `false` | Way(s) for Endpoint to be identified
`mailboxes` | `String` |  | `false` | Mailbox(es) to be associated with
`mohsuggest` | `String` | `default` | `false` | Default Music On Hold class
`outbound_auth` | `Custom` |  | `false` | Authentication object used for outbound requests
`outbound_proxy` | `String` |  | `false` | Proxy through which to send requests
`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_pai` | `Boolean` | `no` | `false` | Send the P-Asserted-Identity header
`send_rpid` | `Boolean` | `no` | `false` | Send the Remote-Party-ID header
`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 transport configuration
`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 interval
`use_avpf` | `Boolean` | `no` | `false` | Determines whether res_pjsip will use and enforce usage of AVPF for this endpoint.
`media_encryption` | `Custom` | `no` | `false` | Determines whether res_pjsip will use and enforce usage of media encryption for this endpoint.
`inband_progress` | `Boolean` | `no` | `false` | Determines whether chan_pjsip will indicate ringing using inband progress.
`callgroup` | `Custom` |  | `false` | The numeric pickup groups for a channel.
`pickupgroup` | `Custom` |  | `false` | The numeric pickup groups that a channel can pickup.
`namedcallgroup` | `Custom` |  | `false` | The named pickup groups for a channel.
`namedpickupgroup` | `Custom` |  | `false` | The named pickup groups that a channel can pickup.
`devicestate_busy_at` | `Unsigned Integer` | `0` | `false` | The number of in-use channels which will cause busy to be returned as device state
`t38udptl` | `Boolean` | `no` | `false` | Whether T.38 UDPTL support is enabled or not
`t38udptl_ec` | `Custom` | `none` | `false` | T.38 UDPTL error correction method
`t38udptl_maxdatagram` | `Unsigned Integer` | `0` | `false` | T.38 UDPTL maximum datagram size
`faxdetect` | `Boolean` | `no` | `false` | Whether CNG tone detection is enabled
`t38udptl_nat` | `Boolean` | `no` | `false` | Whether NAT support is enabled on UDPTL sessions
`t38udptl_ipv6` | `Boolean` | `no` | `false` | Whether IPv6 is used for UDPTL Sessions
`tonezone` | `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.
`recordonfeature` | `String` | `automixmon` | `false` | The feature to enact when one-touch recording is turned on.
`recordofffeature` | `String` | `automixmon` | `false` | The feature to enact when one-touch recording is turned off.
`rtpengine` | `String` | `asterisk` | `false` | Name of the RTP engine to use for channels created for this endpoint
`allowtransfer` | `Boolean` | `yes` | `false` | Determines whether SIP REFER transfers are allowed for this endpoint
`sdpowner` | `String` | `-` | `false` | String placed as the username portion of an SDP origin (o=) line.
`sdpsession` | `String` | `Asterisk` | `false` | String used for the SDP session (s=) line.
`tos_audio` | `Unsigned Integer` | `0` | `false` | DSCP TOS bits for audio streams
`tos_video` | `Unsigned Integer` | `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
`allowsubscribe` | `Boolean` | `yes` | `false` | Determines if endpoint is allowed to initiate subscriptions with Asterisk.
`subminexpiry` | `Unsigned Integer` | `0` | `false` | The minimum allowed expiry time for subscriptions initiated by the endpoint.
`fromuser` | `String` |  | `false` | Username to use in From header for requests to this endpoint.
`mwifromuser` | `String` |  | `false` | Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.
`fromdomain` | `String` |  | `false` | Domain to user in From header for requests to this endpoint.
`dtlsverify` | `Custom` |  | `false` | Verify that the provided peer certificate is valid
`dtlsrekey` | `Custom` |  | `false` | Interval at which to renegotiate the TLS session and rekey the SRTP session
`dtlscertfile` | `Custom` |  | `false` | Path to certificate file to present to peer
`dtlsprivatekey` | `Custom` |  | `false` | Path to private key for certificate file
`dtlscipher` | `Custom` |  | `false` | Cipher to use for DTLS negotiation
`dtlscafile` | `Custom` |  | `false` | Path to certificate authority certificate
`dtlscapath` | `Custom` |  | `false` | Path to a directory containing certificate authority certificates
`dtlssetup` | `Custom` |  | `false` | Whether we are willing to accept connections, connect to the other party, or both.
`srtp_tag_32` | `Boolean` | `no` | `false` | Determines whether 32 byte tags should be used instead of 80 byte tags.

#### 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_screened`
- `allowed_failed_screened`
- `allowed`
- `prohib_not_screened`
- `prohib_passed_screened`
- `prohib_failed_screened`
- `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`

##### dtmfmode

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.

##### identify_by

An endpoint can be identified in multiple ways. Currently, the only supported option is `username`, which matches the endpoint based on the username in the From header.

Note

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 to `username` and an `identify` configuration section exists for the endpoint, then the endpoint can be identified in multiple ways.

- `username`

##### rewrite_contact

On inbound SIP messages from this endpoint, the Contact header will be changed to have the source IP address and port. This option does not affect outbound messages send to this endpoint.

##### timers_min_se

Minimium session timer expiration period. Time in seconds.

##### timers

- `forced`
- `no`
- `required`
- `yes`

##### 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

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

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 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 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.

##### 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.

##### 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.

##### callgroup

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

##### pickupgroup

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

##### namedcallgroup

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

##### namedpickupgroup

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

##### devicestate_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.

##### t38udptl

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

##### t38udptl_ec

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

##### t38udptl_maxdatagram

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

##### faxdetect

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

##### t38udptl_nat

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

##### t38udptl_ipv6

When enabled the UDPTL stack will use IPv6.

##### recordonfeature

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

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

##### recordofffeature

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

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

##### tos_audio

See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings

##### tos_video

See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings

##### cos_audio

See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings

##### cos_video

See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings

##### dtlsverify

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

##### dtlsrekey

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.

##### dtlscertfile

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

##### dtlsprivatekey

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

##### dtlscipher

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

##### dtlscafile

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

##### dtlscapath

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

##### dtlssetup

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.

##### srtp_tag_32

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

### 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 password used for authentication.
`realm` | `String` | `asterisk` | `false` | SIP realm for endpoint
`type` | `None` |  | `false` | Must be 'auth'
`username` | `String` |  | `false` | Username to use for account

#### Configuration Option Descriptions

##### auth_type

This option specifies which of the password style config options should be read when trying to authenticate an endpoint inbound request. If set to `userpass` then we'll read from the 'password' option. For `md5` we'll read from 'md5_cred'.

- `md5`
- `userpass`

##### md5_cred

Only used when auth_type is `md5`.

##### password

Only used when auth_type is `userpass`.

### domain_alias

Domain Alias

#### Configuration Option Reference

Option Name | Type | Default Value | Regular Expression | Description
--- | --- | --- | --- | ---
`type` | `None` |  | `false` | Must be of type 'domain_alias'.
`domain` | `String` |  | `false` | Domain to be aliased

### transport

SIP Transport

#### 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` |  | `false` | File containing a list of certificates to read (TLS ONLY)
`cert_file` | `String` |  | `false` | Certificate file for endpoint (TLS ONLY)
`cipher` | `Custom` |  | `false` | Preferred Cryptography Cipher (TLS ONLY)
`domain` | `String` |  | `false` | Domain the transport comes from
`external_media_address` | `String` |  | `false` | External 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)
`localnet` | `Custom` |  | `false` | Network to consider local (used for NAT purposes).
`password` | `String` |  | `false` | Password required for transport
`privkey_file` | `String` |  | `false` | Private key file (TLS ONLY)
`protocol` | `Custom` | `udp` | `false` | Protocol to use for SIP traffic
`require_client_cert` | `Custom` |  | `false` | Require client certificate (TLS ONLY)
`type` | `None` |  | `false` | Must be of type 'transport'.
`verify_client` | `Custom` |  | `false` | Require verification of client certificate (TLS ONLY)
`verify_server` | `Custom` |  | `false` | Require verification of server certificate (TLS ONLY)
`tos` | `Unsigned Integer` | `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

#### Configuration Option Descriptions

##### cipher

Many options for acceptable ciphers see link for more: http://www.openssl.org/docs/apps/ciphers.html#CIPHER\_STRINGS

##### method

- `default`
- `unspecified`
- `tlsv1`
- `sslv2`
- `sslv3`
- `sslv23`

##### localnet

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

##### protocol

- `udp`
- `tcp`
- `tls`
- `ws`
- `wss`

##### tos

See `https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service` for more information on this parameter.

Note

This option does not apply to the ws or the wss protocols.

##### cos

See `https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service` for more information on this parameter.

Note

This option does not apply to the ws or the wss protocols.

### contact

A way of creating an aliased name to a SIP URI

#### Configuration Option Reference

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

#### Configuration Option Descriptions

##### expiration_time

Time to keep alive a contact. String style specification.

##### qualify_frequency

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

### aor

The configuration for a location of an endpoint

#### Configuration Option Reference

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` | Mailbox(es) to be associated with
`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
`authenticate_qualify` | `Boolean` | `no` | `false` | Authenticates a qualify request if needed

#### Configuration Option Descriptions

##### contact

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

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.

##### maximum_expiration

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

##### max_contacts

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.

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_expiration

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

##### remove_existing

On receiving a new registration to the AoR should it remove the existing contact that was registered against it?

Note

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

##### qualify_frequency

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

##### authenticate_qualify

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

### system

Options that apply to the SIP stack as well as other system-wide settings

#### Configuration Option Reference

Option Name | Type | Default Value | Regular Expression | Description
--- | --- | --- | --- | ---
`timert1` | `Unsigned Integer` | `500` | `false` | Set transaction timer T1 value (milliseconds).
`timerb` | `Unsigned Integer` | `32000` | `false` | Set transaction timer B value (milliseconds).
`compactheaders` | `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` | `false` | Maximum number of threads in the res_pjsip threadpool. A value of 0 indicates no maximum.
`type` | `None` |  | `false` | Must be of type 'system'.

#### Configuration Option Descriptions

##### timert1

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.

##### timerb

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.

### global

Options that apply globally to all SIP communications

#### Configuration Option Reference

Option Name | Type | Default Value | Regular Expression | Description
--- | --- | --- | --- | ---
`maxforwards` | `Unsigned Integer` | `70` | `false` | Value used in Max-Forwards header for SIP requests.
`type` | `None` |  | `false` | Must be of type 'global'.
`useragent` | `String` | `Asterisk PBX SVN-branch-12-r397924` | `false` | Value used in User-Agent header for SIP requests and Server header for SIP responses.

### Import Version

This documentation was imported from Asterisk Version Unknown