Configuring DWG2000C Dinstar Hope it works for you

DWG2000C DINSTAR GATEWAY CONFIGURATION - ELASTIX

versions used: Asterisk Server: Elastix 2.0.0-63 IP: 192.168.150.2
Gateway GSM: 2.22.04.02 firmware DWG2000C DinStar IP: 192.168150.8
It is assumed that you have a Asterisk server working properly. Some Fields I left empty here are also empty in
the configuration.
CONFIGURATION ELASTIX
It creates a new SIP trunk pointing GSM Gateway (PBX> Trunks> AddTrunk> Add SIP Trunk)
Trunk Description: Celular_DWG_1
Trunk Name : Celular_DWG_1
PEER Details: host = 192.168.150.8
context = from-trunk
type = peer
insecure = port, invite
changes are saved (Submit Changes, Apply Configuration)
Creates a new Outbound Route dial using the desired pattern (in Colombia 3XXXXXXXXX ) towards the trunk
previously created (PBX> Outbound Routes> Add Route)
Route Name: Celular_DWG_1
PIN Set: Cellular
Dial Patterns: 3XXXXXXXXX
Trunk Sequence: 0 SIP/Celular_DWG_1
changes are saved (Submit Changes, Apply Configuration)
It creates a new Inbound Route to route incoming calls, in our case will be routed to the IVR Welcome, will be
used to identify the CID 4858999 incoming calls (PBX> Inbound Routes> Add Incoming Route)
DID Number: 4858999
Set Destination: IVR:
Welcome are saved changes (Submit Changes, Apply Configuration)
DWG2000C DINSTAR GATEWAY CONFIGURATION
Configure the IP address of the Gateway DinStar (Network Configuration> Local network)
Use the following IP address IP Address: 192.168.150.8
Subnet Mask: 255.255.255.0
Default Gateway: 192.168. 150.2 (Note that this is the IP address of Elastix server)
IP Trunk is created pointing to our Asterisk server (IP Trunk Configuration> IP Trunk> Add)
Index: 31
IP: 192.168.150.2
Port: 5060
Description: COLAST01
Enable KeepAlive:

On mode is configured routing (Routing Configuration> Routing Parameter)

IP-> Tel Parameter: Route calls before manipulation
Tel-> IP parameter: Route calls before manipulation
is configured for routing calls from Elastix to Cell (Routing Configuration> IP- > Call Routing> Add)
Index: 0
Description: default
Source Prefix: any
Source IP: 31 <COLAST01>
Destination Prefix: any
Port Group: 0 <all>
is configured for routing calls from Mobile to Elastix (Routing Configuration> Tel- > IP Routing> Add)
Index: 0
Description: default
Source Prefix: any
Source Port: Any
Destination Prefix: any
Destination: IP: 31 <COLAST01>
permissions are configured for calls from Elastix to Cell (Operation> IP-> Tel Operation)
Index: 31
Source Prefix: any
Source IP: IP: 31 <COLAST01>
Destination Prefix: any
Operation: Allow
Call Description: COLAST01_Out
permissions are configured for calls from cell phones to Elastix (Operation> Tel-> IP Operation)
Index: 31
Source Prefix: any
Source Port: Port: Any
Destination Prefix: any
Operation: Allow Call
Auto Call: Enable
Description: COLAST01_In
Configures service settings (System Configuration> Service Parameter)
Start Local RTP Port: 8000
Enable silence suppression : Yes
Call Progress Tone: USA
Preferred Coders: PCMU, PCMA, G729AB, G723.1
Voice Frames per TX: 2
Do Not Answer Incoming Call for GSM Hotline: Yes
Enable Incoming GSM Configuration: Yes
Auto Routing Outgoing Type: Polling
IP to GSM One Stage Dialing: Yes
Answer Delay: 5
Call Redirect When All Ports Busy: No
Play Voice Prompt for incoming GSM Calls: Yes
Parameter DTMF
DTMF Method: RFC2833

RFC2833 Payload Type: 101

DTMF Volume: 0dB
DTMF interval: 200
NAT Transversal: Disable
Other Configuration
Enable Private Service: Yes
User ID is Phone Number: Yes
Only Accept Calls from SIP Server: No
Allow Call from GSM to IP without Registration: Yes
Allow Call from IP to GSM without Registration: Yes
Reject Anonymous call from IP to GSM: No
Use # as End Key: Yes
No Answer TimeOut: 55
interdigit timeout: 4
Call Delay: 0
are configured SIP account settings (System Configuration> Parameter SIP)
SIP Proxy
SIP Server Address: 192.168.150.2
SIP Server Port: 5060
Check Status: No
Outbound Proxy Address
Outbound Proxy:
Outbound Proxy Port: 5060
Is Register: No
DNS Query type: A query
DNS refresh interval: 0
T1: 500
T2: 4000
T4: 5000
TMAX: 32000
Keepalive Interval: 32
Enable 100rel: Not
From When Caller ID mode is Available: Tel / Tel
mode From Caller ID is Unavailable When: Username
Answer Mode: Answered 183 Mode: Immediately
Session timer: No
Parameters are configured ports (System Configuration> Port Parameter> Detail)
All Ports Register same user ID used: Yes
SIP User ID:
Authenticate ID:
Authenticate Password:
Local SIP Port: 5060
Tx Gain: +2 dB
Rx Gain: +6 dB To VOIP Hotline: 4858999 (This is the number we sent to the inbound route Elastix
To PSTN Hotline: Auto-Dial Delay Time. 0
With this we have completed the configuration, restart the Gateway now and are tested for incoming and
outgoing calls