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Table of Contents

 

Table of Contents

1.  Overview

What is DGW-100XR?

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     Table 4-1-1 Definition of SIP Options

 

Options

Definition

Name

A name which is able to read by human. And it’s only used for user’s reference.

Username

User name the end point use to authenticate with the gateway

Password

Password the endpoint will use to authenticate with the gateway. Allowed characters

Registration

Whether this endpoint will registers with this gateway.

Hostname or IP Address

IP address or hostname of the endpoint or 'dynamic' if the endpoint has a dynamic IP address. This will require registration. Notice: if the input here is hostname and your DNS has changed, you must reboot asterisk.

Transport

This sets the possible transport types for outgoing. Order of usage, when the respective transport protocols are enabled, is UDP, TCP, TLS. The first enabled transport type is only used for outbound messages until a Registration takes place. During the peer Registration the transport type may change to another supported type if the peer requests so.

NAT Traversal

Addresses NAT-related issues in incoming SIP or media sessions.

 

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          Table 4-1-2 Definition of Registration Options

 

Options

Definition

Authentication User

A username to use only for registration.

Register Extension

When Gateway registers as a SIP user agent to a SIP proxy (provider), calls from this provider connect to this local extension.

From User

A username to identify the gateway to this endpoint.

From Domain

A domain to identify the gateway to this endpoint.

Remote Secret

A password which is only used if the gateway registers to the remote side.

Port

The port number the gateway will connect to at this endpoint.

Qualify

Whether or not to check the endpoint's connection status.

Qualify frequency Frequency

How often, in seconds, to check the endpoint's connection status.

Outbound Proxy

A proxy to which the gateway will send all outbound signaling instead of sending signaling directly to endpoints.

 

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                              Table 4-1-3 Definition of Call Options

 

Options

Definition

DTMF Mode

Set default DTMF Mode for sending DTMF. Default: rfc2833. 
Other options: 'info', SIP INFO message (application/ dtmf-relay);
'Inband', Inband audio (require 64kbit codec - alaw, ulaw).

Trust Remote-Party-ID

Whether or not the Remote-Party-ID header should be trusted.

Send Remote-Party-ID

Whether or not to send the Remote-Party-ID header.

Caller ID Presentation

Whether or not to display Caller ID.

 

Advanced Timer Settings

 

                      Table 4-1-4 Definition of Timer Options

 

Options

Definition

Default T1 Timer

This timer is used primarily in INVITE transactions. The default for Timer T1 is 500ms or the measured run-trip time between the gateway and the device if you have qualify=yes for the device.

Call Setup Timer

If a provisional response is not received in this amount of time, the call will auto-congest. Defaults to 64 times the default T1 timer. 

Session Timers

Session-Timers feature operates in the following three modes: originate, Request and run session-timers always; accept, run session-timers only when requested by other UA; refuse, do not run session timers in any case.

Minimum Session

Minimum session refresh interval in seconds. Default is 90secs.

Maximum Session Refresh Interval

Maximum session refresh interval in seconds. Defaults to 1800s.

Session Refresher

The session refresher, uac or uas. Defaults to uas.

 

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                                Table 4-2-1 Definition of Networking Options

 

Options

Definition

UDP Bind Port

Choose a port on which to listen for UDP traffic.

Enable TCP

Enable server for incoming TCP connection (default is no).

TCP Bind Port

Choose a port on which to listen for TCP traffic.

TCP Authentication Timeout

The maximum number of seconds a client has to authenticate. If the client does not authenticate before this timeout expires, the client will be disconnected.(default value is: 30 seconds).

TCP Authentication Limit

The maximum number of unauthenticated sessions that will be
allowed to connect at any given time (default is: 50).

Enable Hostname Lookup

Enable DNS SRV lookups on outbound calls Note: the gateway only uses the first host in SRV records Disabling DNS SRV lookups disables the ability to place SIP calls based on domain names to some other SIP users on the Internet specifying a port in a SIP peer definition or when dialing outbound calls with suppress SRV lookups for that peer or call.

Enable Internal SIP Call

Whether enable the internal SIP calls or not when you select the registration option "Endpoint registers with this gateway".

Internal SIP Call Prefix

Specify a prefix before routing the internal calls.

 

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                        Table 4-2-2 Definition of NAT Settings Options

 

Options

Definition

Local Network

Format:192.168.0.0/255.255.0.0 or 172.16.0.0./12. A list of IP address or IP ranges which are located inside a NATed network. This gateway will replace the internal IP address in SIP and SDP messages with the external IP address when a NAT exists between the gateway and other endpoints.

Local Network List

Local IP address list that you added.

Subscribe Network Change Event

Through the use of the test_stun_monitor module, the gateway has the ability to detect when the perceived external network address has changed. When the stun_ monitor is installed and configured, chan_sip will renew all outbound registrations when the monitor detects any sort of network change has occurred. By default this option is enabled, but only takes effect once res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not generate all outbound registrations on a network change, use the option below to disable this feature.

Match External Address Locally

Only substitute the exeternaddr or externhost setting if it matches

Dynamic Exclude Static

Disallow all dynamic hosts from registering as any IP address used for staticly defined hosts .This helps avoid the configuration error of allowing your users to register at the same address as a SIP provide.

Externally Mapped TCP Port

The externally mapped TCP port, when the gateway is behind a static NAT or PAI

External Address

The external address (and optional TCP port) of the NAT. External address=hostname [:port] specifies a static address[:port] to be used in SIP and SDP messages. Examples: External address=12.34.56.78 External address=12.34.56.78.9900

 

 

 

External Hostname

The external hostname (and optional TCP port) of the NAT.

External Hostname=hostname[:port] is similar to

“External address”. Examples:

External Hostname=foo.dyndns.net

Hostname Refresh Interval

How often to perform a hostname lookup. This can be useful when your NAT device lets you choose the port mapping, but the IP address is dynamic. Beware, you might suffer from service disruption when the name server resolution fails.

 

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              Table 4-2-3 Definition of RTP Settings Options

 

Options

Definition

Start of RTP Port Range

Start of range of port numbers to be used for RTP.

End of RTP port Range

End of range of port numbers to be used for RTP.

 

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                      Table 4-2-4 Instruction of Parsing and Compatibility

 

Options

Definition

Strict RFC Interpretation

Check header tags, character conversion in URIs, and multiline headers for strict SIP compatibility(default is yes)

Send Compact Headers

Send compact SIP headers

SDP Owner

Allows you to change the username filed in the SDP owner string.
This filed MUST NOT contain spaces.

Disallowed SIP Methods

When a dialog is started with another SIP endpoint, the other endpoint should include an Allow header telling us what SIP methods the endpoint implements. However, some endpoint either do not include an Allow header or lie about what methods they implement. In the former case, the gateway makes the assumption that the endpoint support all known SIP methods. If you know that your SIP endpoint does not provide support for a specific method, then you may provide a list of methods that your endpoint does not implement in the disallowed_ methods option. Note that if your endpoint is truthful with its Allow header, then there is need to set this option.

Shrink Caller ID

The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not in square brackets. For example, the caller id value 555.5555 becomes 5555555 when this option is enabled. Disabling this option results in no modification of the caller id value, which is necessary when the caller id represents something that must be preserved. By default this option is on.

Maximum Registration Expiry

Maximum allowed time of incoming registrations and subscriptions (seconds).

Minimum Registration Expiry

Minimum length of registrations/subscriptions (default 60).

 

 

Default Registration Expiry

 

Default length of incoming/outgoing registration.

Registration Timeout

How often, in seconds, to retry registration calls. Default 20 seconds.

Number of Registration

Number of registration attempts before we give up.0=continue forever, hammering the other server until it accepts the registration. Default is 0 tries, continue forever.

 

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                       Table 4-2-5 Instruction of Security

 

Option

Definition

Match Auth Username

If available, match user entry using the 'username' field from the
authentication line instead of the 'from' field.

Realm

Realm for digest authentication. Realms MUST be globally unique according to RFC 3261. Set this to your host name or domain name.

Use Domain as Realm

Use the domain from the SIP Domains setting as the realm. In this case, the realm will be based on the request 'to' or 'from' header and should match one of the domain. Otherwise, the configured 'realm' value will be used.

Always Auth Reject

When an incoming INVITE or REGISTER is to be rejected, for any reason, always reject with an identical response equivalent to valid username and invalid password/hash instead of letting the requester know whether there was a matching user or peer for their request. This reduces the ability of an attacker to scan for valid SIP usernames. This option is set to 'yes' by default.

Authenticate Options Requests

Enabling this option will authenticate OPTIONS requests just like INVITE requests are. By default this option is disabled.

Allow Guest Calling

Allow or reject guest calls (default is yes, to allow). If your gateway is connected to the Internet and you allow guest calls, you want to check which services you offer everyone out there, by enabling them in the default context.

 

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          Table 4-2-6 Instruction of Media

 

Options

Definition

TOS for SIP Packets

Sets type of service for SIP packets

TOS for RTP Packets

Sets type of service for RTP packets

 

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                     Table 5-1-1 Definition of Routing Options

 

Options

Definition

Routing Name

The name of this route. Should be used to describe what types of calls this route matches (for example, 'SIP2Ports' or 'Ports2SIP').

Call Comes in From

The launching point of incoming calls.

Send call Through

The destination to receive the incoming calls.

 

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                           Table 5-1-2 Description of Advanced Routing Rule

 

Options

Definition

Dial Patterns that will use this Route

A Dial Pattern is a unique set of digits that will select this route and send the call to the designated trunks. If a dialed pattern matches this route, no subsequent routes will be tried. If Time Groups are enabled, subsequent routes will be checked for matches outside of the designated time(s).
Rules:
X matches any digit from 0-9
  matches any digit from 1-9
matches any digit from 2-9
[1237-9]  matches any digit in the brackets (example: 1,2,3,7,8,9)
wildcard: matches one or more dialed digits.
prepend:  Digits to prepend to a successful match.
If the dialed number matches the patterns specified by the subsequent columns, then this will be prepended before sending to the trunks.

prefix:   Prefix to remove on a successful match.
The dialed number is compared to this and the subsequent columns for a match.
Upon a match, this prefix is removed from the dialed number before sending it to the trunks.

match pattern:   The dialed number will be compared against the prefix + this match pattern.
Upon a match, the match pattern portion of the dialed number will be sent to the trunks

SDfR(Stripped Digits from Right): The amount of digits to be deleted from the right end of the number. If the value of this item exceeds the length of the current number, the whole number will be deleted.

RDfR( Reserved Digits from Right) :Designated information to be added to the right end of the current number.

StA(Suffix to Add):Designated information to be added to the right end of the current number.

Caller Name: What caller name would you like to set before sending this call to the endpoint. Native language charset is allowable, e.g. Chinese charset, Latin charset.

Forward Number

What destination number will you dial? 

This is very useful when you have a transfer call.

Failover Call Through Number

The gateway will attempt to send the call out each of these in the order you specify.

 

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             Table 6-1-1Definition of WAN/LAN Settings

 

Options

Definition

Interface

The name of network interface.

Type

The method to get IP.

Static: manually set up your gateway IP.

DHCP: automatically get IP from your local LAN.

MAC

Physical address of your network interface.

Address

The IP address of your gateway.

Network

The subnet mask of your gateway.

Default Gateway

Default getaway IP address.

 

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         Table 6-2-1 Definition of DDNS Settings

 

Options

Definition

DDNS

Enable/Disable DDNS(dynamic domain name server)

Type

Set the type of DDNS server.

Username

Your DDNS account’s login name.

Password

Your DDNS account’s password.

Your domain

The domain to which your web server will belong.

 

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               Table 7-1-1 Definition of Asterisk API

 

Options

Definition

Port

Network port number

Manager Name

Name of the manager without space

Manager secret

Password for the manager.

Characters: Allowed characters “-_+.<>&0-9a-zA-Z”. Length:4-32 characters.

Deny

If you want to deny many hosts or networks, use char & as separator. Example: 0.0.0.0/0.0.0.0 or 192.168.1.0/255.255.255.0&10.0.0.0/255.0.0.0

Permit

If you want to permit many hosts or network, use char & as separator. Example: 0.0.0.0/0.0.0.0 or 192.168.1.0/255.255.255.0&10.0.0.0/255.0.0.0

System

General information about the system and ability to run system management commands, such as Shutdown, Restart, and Reload.

Call

Information about channels and ability to set information in a running channel.

Log

Logging information. Read-only. (Defined but not yet used.)

Verbose

Verbose information. Read-only. (Defined but not yet used.)

Command

Permission to run CLI commands. Write-only.

Agent

Information about queues and agents and ability to add queue members to a queue.

User

Permission to send and receive UserEvent.

Config

Ability to read and write configuration files.

DTMF

Receive DTMF events. Read-only.

Reporting

Ability to get information about the system.

Dialplan

Receive NewExten and Var Set events.  Read-only.

Originate

Permission to originate new calls. Write-only.

All

Select all or deselect all.

 

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             Table 7-2-1 Definition of Asterisk CLI

 

Options

Definition

Command

Type your Asterisk CLI commands here to check or debug your gateway.

 

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                          Table 8-1-1 Definition of Logs

 

Options

Definition

Auto clean:

(System Logs)

switch on : when the size of log file reaches the max size,

the system will cut a half of the file. New logs will be retained.

switch off : logs will remain, and the file size will increase gradually.

default on, default size=1MB

Verbose:

Asterisk console verbose message switch.

Notice:

Asterisk console notice message switch.

Warning:

Asterisk console warning message switch.

Debug:

Asterisk console debug message switch.

Error:

Asterisk console error message switch.

DTMF:

Asterisk console DTMF info switch.

Auto clean:

(asterisk logs)

switch on : when the size of log file reaches the max size,

the system will cut a half of the file. New logs will be retained.

Switch off: logs will remain, and the file size will increase gradually.

default on, default size=100KB

SIP Logs:

Whether enable or disable SIP log.

Auto clean:

(SIP logs)

switch on : when the size of log file reaches the max size,

the system will cut a half of the file. New logs will be retained.

Switch off: logs will remain, and the file size will increase gradually.

default on, default size=2MB

IAX2 Logs

Whether enable or disable IAX log

Auto clean

switch on : when the size of log file reaches the max size,

the system will cut a half of the file. New logs will be retained.

Switch off: logs will remain, and the file size will increase gradually.

default on, default size=2MB

MFC/ R2 Logs

Whether enable or disable MFC/ R2 Logs log.

Auto clean

switch on : when the size of log file reaches the max size,

the system will cut a half of the file. New logs will be retained.

Switch off: logs will remain, and the file size will increase gradually.

default on, default size=100KB

PRI Logs

PRI port logs. You can choose one or more ports. If you choose "All", the "PRI" page will show you the logs about all the ports.

Auto clean (PRI logs)

switch on : when the size of log file reaches the max size,

the system will cut a half of the file. New logs will be retained.

Switch off: logs will remain, and the file size will increase gradually.

default on, default size=2MB

.SS7 Logs

Whether enable or disable SS7 log

Auto clean

switch on : when the size of log file reaches the max size,

the system will cut a half of the file. New logs will be retained.

Switch off: logs will remain, and the file size will increase gradually.

default on, default size=100KB

Call Statistics

Whether enable or disable Call Statistics.

   

 

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