NEC-SIP trunks - NETSIP

NEC-SIP trunks - NETSIP


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NETSIP-  SIP Trunks

Pre-requisite:

SIP ALG disabled on the customers router

Method:

NETSIP servers use one of many IP addresses, you register to one, however calls may come from another , causing the call to get blocked by your on site router. UDP port 5060 needs to be port forwarded to your NEC, A router must support an allow list from the following 3 x C Classes

103.26.173.0/24

103.26.174.0/24

103.26.175.0/24


DO NOT PORT FORWARD FROM ANY PUBLIC IP ADDRESS,  use allow lists!!


NETSIP use an account number to REGISTER , however INVITES must contain a valid 10 digit CLIP. By default all carrier selections ( 10-29-14 , A-Z ) use the Account “User ID” in 10-36-xx series to REGISTER it also uses the same “User ID” in the From: field in the INVITE , we must change the From: field in the INVITE to a valid 10 digit CLIP in order to make a successful out bound call

This 84-39 series can only be modified via PC-Pro in the way of scripts. ( SL2100 you can manually change them in PCPro with our using scripts )








In order to modify these values you must:


1. Download the system configuration 

2. Run the Script 



you must use “SIP Script” if you are using profiles 2 tru 6. 99% of installs you will only use profile 1, If you are using profile 1 then you can select either Run script or Run SIP Script)

Script attached at the end of this article.


3.  Complete the SIP trunk programming.

4.  Upload the database

Hardware / License required:


SV9100:


Quantity

Description

SV9100 CPU

1

System

GPZ-IPLE

1

VoIP Media card to convert RTP packets

BE114065-CP20

Per SIP trunk

SV9100 IP trunk License


BE114065-CP20 appears as a 500lic on the system:



SL2100:


Quantity

Description

SL2100 System

1

System

IP7WW-VOIPDB-C1

?

VoIP Media card to convert RTP packets

BE116745

Per IP phone

IP Phone License

Note: for the SV2100 VOIPDB is only required if you need to exceed 8 VoIP channels


BE116745 appears as a 500Lic on the system

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System Programming:

Check in your License List for enough VoIP resources. 1 per trunk for Non IP terminal systems, 2 per trunk for IP terminal systems.

NOTE:

For SV9100’s with more than 8 VoIP resources required, make sure you check :

If the Licenses do not equal you License shown in the Licence area you need to subtract 8 & enter the result in 10-54 as well as the License code 5103

PRG 10-12-09    Valid IP address on the customers network

RRG 10-12-10    Subnet mask that matches the customers network

PRG 84-26-01    Valid IP address on the customers network ( by convention, one IP above 10-12-09 )

PRG 10-68-01    SIP trunk Type / Start port & number of ports

Be careful this doesn’t conflict with any COT / ISDN trunk cards

10-67-01 Turn DNS mode on

10-67-02 enter a valid DNS server

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21-17  Add CLIP to lines assigned ( 10 digit format)



PBX Configuration Guide on Netsip Service:



General SIP Settings

• SIP 2.0 via UDP or TCP

• RTP via UDP

• DTMF RFC2833

• SIP registration is supported, we suggest expires of 60 minutes when using a static IP address.

SIP Servers

If your PBX supports DNS SRV record, or are happy with only pointing at 1 server.

• sip.netsip.net.au

If your PBX's can't support SRV records, you can manually create outbound routes to the below servers.

• 100wic.qld.sip.netsip.net.au

• 24lit.qld.sip.netsip.net.au

• 47bou.nsw.sip.netsip.net.au

• 400har.nsw.sip.netsip.net.au

• 826lor.vic.sip.netsip.net.au

Firewall Settings

Please allow SIP (TCP/UDP 5060), RTP (UDP 16000-65000) & ICMP (For monitoring/trouble shooting) to/from the

following IP subnets:

• 103.26.173.0/24

• 103.26.174.0/24

• 103.26.175.0/24

 

 

Caller ID

All calls need to be presented with a valid phone number (10 Digit 0NSN, or E164 format) as CallerID for all calls.

We support the P-Asserted Identity with the Privacy field (RFC3325), the SIP From field and Remote Party ID as

sources of CallerID.

Our CallerID format preference:

• PAI (P-Asserted Identity - RFC3325)

• Remote Party ID

• From field

 

QOS

NetSIP voice switches set the following QOS markings, we prefer customer to use these values. Our Network will honour the below values

Type DSCP Decimal

SIP Signalling CS3 24

RTP Audio EF 46

 

Codecs

We support the below codec's on our voice network

Preference Codec khz ptime

1 G722 8 / 16 20ms

2 G711a 8 20ms

3 G729 a&b 8 20ms

4 G711u 8 20ms

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