SIP TRUNKS

[Chamvari]

thread1613-1492098

I have searched for several threads including the one above for SIP Trunk programming and have gone to the manual itself but looks like i am not having any success for the past 4 months. I need to program SIP trunks on IPLdk 100 . Sometimes i can register to the server sometimes i cant make calls with speech.
Is there a step by step programming for this program, i would really appreciate if anyone can help i can see it has been discussed on this forum and there were some data base exchange of which i don't have conclusive info for the final programming.

Dial Tone shoul be There

 

[SteveBUK]

Are you using IP Address Authenticaion
OR
User Name and Password Authentication

 

[SteveBUK]

IP Address Auth Programming
Must check MPB and VOIBE software
ipLDK 50/100/300 - 3.7Ce MPB Software or above
VOIBE - 2.1Cb Software or above

In this example :
SIP provider has supplied a range of 10 DDI numbers, these are 01403 330320
to 01403 330329.
The SIP Proxy Server Port(Socket) is 5060.
The codec being used is G711 A_Law (as first choice)
4 SIP trunks have been provided by provider.

The Router is using NAT.
The ADSL being used is ADSL Max.
Bandwidth = upstream 500Kbps, downstream 5Mbps
Contention Ratio = 20:1
The Router has been programmed to open (forward) ports between the
external side (217.37.128.90) and the internal VOIP address
(192.168.10.4)
The ports opened are as follows :
Programming Example
TCP 1719 Call set up / control
TCP 1720 Call set up / control
UDP 2048 to 4096 RTP voice transport
UDP 5060 Gamma Server Port (Socket)

PGM 340 - VOIB/Gatekeeper
IP Address - 192.168.10.4 (ip address of the VOIBE)
GATEWAY Address - 192.168.10.253 (Internal address of Router)
SUBNET Mask - 255.255.255.0 (Subnet mask of LAN)
Default Codec - G.711 A_Law (Set as required)
Firewall IP Address - 217.37.128.90 External (forwarded) address of the Router
VOIB Mode SIP/H323/Dual If using both SIP and H323 set to Dual note - ipNetworking and ipEndpoints (hardphone/softphone) use H323)



PGM 341 - VOIB/Gatekeeper
SIP Attribute 1
Proxy Server Address- 83.245.6.81 (Supplied by SIP provider)
Proxy Server Port 5060 (Supplied by SIP Provider)
Proxy Registration Timer 1800
Use Outbound Proxy ON
Primary DNS Address 192.168.10.253 (Internal address of Router)
Secondary DNS Address - blank
Domain 83.245.6.81 (SIP Provider Proxy Server Address)
Connection Mode UDP
100Rel Support OFF
Use Rport Method OFF
Use Single Code Only OFF
Remote Party ID OFF
181 Message ON
IP Centrex OFF
SIP Name Service OFF

PGM 140 -CO Line List
Type ISDN DID (Set for DDI working)

PGM 322 Networking CO Line Attribute
VOIP Mode SIP (Set for SIP working)

PGM 146 CO ISDN Attr
Incomming Prefix Code Insertion ON (If SIP Provider does not provide the leading zero)
Outgoing Prefix Code Insertion OFF

PGM 143 CO ISDN Attr
DDI Converstion Type 2 (For Flex DDI)
ISDN ENBLK Send ON

Sending CLI
PGM 200 ISDN Attributes
My Area Code 1403 (STD code without leading 0)
My Area Prefix Code 0 (To insert the leading 0)

PGM 201 COLP Table
Index 0 - 330 (To add the first 3 digits of the local number)

PGM 143 CO ISDN Attr
COLP Table Idx - 0 (To use index 0)
CLIP Table Idx - 0 (To use index 0)
Call Type - National

PGM 114 STA ISDN Attr
ISDN CLI Station - 320 (To send the last 3 digits of the CLI)

PGM 231 - to route DDI as needed.

Regards
Steve

 

[Chamvari]

and it works thanks i finally got, the problem is our isp does nt support flexible numbering one from their Server we had to customise the existing numbering plan to suit their SIP line Identity.



Dial Tone shoul be There