Table of Contents
Table of Contents |
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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. |
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 |
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
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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 | ||
| Check header tags, character conversion in URIs, and multiline headers for strict SIP compatibility(default is yes) | ||
| Send compact SIP headers | ||
| Allows you to change the username filed in the SDP owner string. | ||
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 allowed time of incoming registrations and subscriptions (seconds). | ||
Minimum Registration Expiry | Minimum length of registrations/subscriptions (default 60). | ||
| 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 | |
| If available, match user entry using the 'username' field from the | |
| Realm for digest authentication. Realms MUST be globally unique according to RFC 3261. Set this to your host name or domain name. | |
| 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. | |
| 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 | |
| Sets type of service for SIP 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). prefix: Prefix to remove on a successful match. match pattern: The dialed number will be compared against the prefix + this match pattern. 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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