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3. source strings
4. review

Default

Default Context

Default TLS Port Assignment

Default bind port for CHAN_PJSIP is: %s, CHAN_SIP is: %s

Default context for incoming calls if not specified. FreePBX sets this to from-sip-external which is used in conjunction with the Allow Anonymous SIP calls. If you change this you will effect that behavior. It is recommended to leave this blank.

Default length of incoming and outgoing registrations.

Detect Network Settings

Disabled

Domain the transport comes from

Don't Require verification of server certificate (TLS ONLY).

Don't Verify Server

Dynamic Host

Dynamic Host Refresh

Dynamic Host can not be blank

Dynamic IP

ERRORS

Edit Settings

Enable Jitter Buffer

Enable SRV Lookup

Enable TCP

Enable TLS

Enable server for incoming TLS (secure) connections.

Enabled

Enables jitter buffer frame logging.

Enables the use of a jitterbuffer on the receiving side of a SIP channel. An enabled jitterbuffer will be used only if the sending side can create and the receiving side can not accept jitter. The SIP channel can accept jitter, thus a jitterbuffer on the receive SIP side will be used only if it is forced and enabled. An example is if receiving from a jittery channel to voicemail, the jitter buffer will be used if enabled. However, it will not be used when sending to a SIP endpoint since they usually have their own jitter buffers. See jbforce to force its use always.

End

Endpoint Identifier Order

Endpoint Identifier Order . The Default order is as follows:<ul><li>ip</li><li>username</li><li>anonymous</li><li>header</li><li>auth_username</li></ul><ul>Note : Changing this to get affected may require asterisk restart</ul>

Enter IP Address

Error: %s

External Address

External FQDN as seen on the WAN side of the router and updated dynamically, e.g. mydomain.example.com. (asterisk: externhost)

External IP Address

External IP can not be blank when NAT Mode is set to Static and no default IP address provided on the main page

External Static IP or FQDN as seen on the WAN side of the router. (asterisk: externip)

Fixed

Force Jitter Buffer

Forces the use of a jitterbuffer on the receive side of a SIP channel. Normally the jitter buffer will not be used if receiving a jittery channel but sending it off to another channel such as another SIP channel to an endpoint, since there is typically a jitter buffer at the far end. This will force the use of the jitter buffer before sending the stream on. This is not typically desired as it adds additional latency into the stream.

Frequency in seconds to check if MWI state has changed and inform peers.

General SIP Settings

Generate manager events when sip ua performs events (e.g. hold).

Helpful Information

Hostname or address for the STUN server used when determining the external IP address and port an RTP session can be reached at. The port number is optional. If omitted the default value of 3478 will be used. This option is blank by default. (A list of STUN servers: http://wiki.freepbx.org/x/YQCUAg)

Hostname or address for the TURN server to be used as a relay. The port number is optional. If omitted the default value of 3478 will be used. This option is blank by default.

ICE Blacklist

ICE Host Candidates

IMPORTANT: Only use this functionality when your Asterisk server is behind a one-to-one NAT and you know what you're doing. If you do define anything here, you almost certainly will NOT want to specify 'stunaddr' or 'turnaddr' above.

IP Addresses

IP Configuration

If blank, will use the default settings