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How to Configure SPA3102 as SIP Trunk on Elastix

 

 

http://www.gkhan.in/configure-spa3102-sip-trunk-elastix/

 

 

How to Configure SPA3102 as SIP Trunk on Elastix or How to configure Elastix PBX SIP Trunk for SPA3102.
If you want to use SPA 3102 as voice gateway with Elastix PBX .this is the step by step guide to configure Elastix  PBX and SPA3102.diagram

1 – Add and Configure the SIP trunk on Elastix Server for SPA3102.

Go to PBX > Trunks > Add SIP Trunk

2- Under General Setting

genrelsetting

Trunk name:-   1-pstn - suggest to use PHONE NUMBER here

outbound caller id: same than trunk name

Maximum Channels : 1

 

 

DialNumber

 

4- Under Outgoing Settings

Dialout

Note :- Trunk Name should be same as above you have given in General Setting.

 

disallow=all
allow=ulaw
canreinvite=no
context=from-trunk
dtmfmode=rfc2833
host=dynamic
incominglimit=1
nat=never
port=5061
qualify=yes
secret=PHONENUMBER
type=friend
username=PHONENUMBER

 

 

Note :- username should be same as trunk name otherwise your SPA3102 will not register with  Elastix .

5- Under incoming setting  leave all blank

incomming

registration

Now Submit changes and apply the changes

 

6- Configure SPA3102 For SIP Trunk

 

On the WAN Setup page configure the Static IP address subnet mask and gateway DNS.

WAN_Setting

On the LAN setup page

choose the networking services and Bridge

LAN_Setting

So the LAN and WAN port will be bridge So you no need to worry about Ethernet connection.

PSTN_NAT

Under NAT Setting

Choose NAT mapping enable  NO

SIP Port must be 5061

sip_setting

PSTN_Proxy

Proxy:  IP address of your Asterisk box

Make Call Without Reg: Yes
Ans Call Without Reg: Yes

Register Expires: 300

PSTN_subscriber

User ID: 1-pstn

Password :-qwerty123

Note:-very important – User-ID  must exactly match the  Trunk name and username in the trunk configuration.

PSTN_audio

 

DTMF Process INFO: Yes
DTMF Process AVT: Yes
DTMF Tx Method: Auto

 

dialplan

 

Don’t change anything in dial plan except Dial Plan 2

Replace 5551234 with the telephone number of the PSTN line coming into the device. Note that this must exactly match the DID number in your FreePBX Inbound Route setting for this device. If the number here and in the Inbound Route don’t match exactly, you won’t receive incoming calls.

 

VOIP_To_PSTN

 

-To-PSTN Gateway Enable: yes
VoIP Caller Auth Method: None 
VoIP PIN Max Retry: 3 
One Stage Dialing: Yes 
Line 1 VoIP Caller DP: none
VoIP Caller Default DP: none
Line 1 Fallback DP: none

 

PSTN_to_voip

 

 

PSTN-To-VoIP Gateway Enable: Yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: no ;
PSTN Pin Max Retry: 3
PSTN CID for VoIP CID: Yes
PSTN CID Number Prefix: (Leave Blank)
PSTN Caller Default DP: 2
Off Hook While Calling VoIP: No
Line 1 Signal Hook Flash To PSTN: Disabled
PSTN CID Name Prefix: (Leave Blank)

Leave everything else in this section blank. We are almost finished now.

 

 

fxo_timer_value

 

 

Voip Answer Delay: 0 

PSTN Answer Delay:3

I got a good article which i use for Elastix SIP Trunk testing from where I configure and test above configuration.

 

Now make OUT Going Route & Incoming call Route

 

 

 

If PSTN italian lines makes lot of echo, can adjust this way: http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/spa3102-voice-gateway-router/108731-pqa-108731.html

 

 

 

 

 

http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx