configuration sipura 3000

[sip_quest]

Hi Folks,
I am very new to sipura 3000. i want to make a voip to pstn call. i want to make a pstn call from SJ phone through sipura 3000. In the SJ phone i configured our local Proxy Ip address. whenever i make a call, it hits the proxy, proxy route the call to sipura 3000. proxy is successfully route the call to sipura 3000.

SJ phone ---(my mobile number)--------> proxy ----> sipura3000 -------------> my mobile

after receiving the INVITE, sipura responds with 100 , 180 , 200 OK. but my mobile is not ringing. but one IVR is playing like callee number is not reachable. callee number is correct only

Note : Am trying this scenario without registration. PSTN subscription name, password is empty

Line Enable: NO

PSTN Line : yes
Preferred Codec: G711u


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


PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: yes
PSTN Caller Auth Method: none
PIN PSTN Ring Thru Line 1: no
PSTN PIN Max Retry: 3
PSTN CID For VoIP CID: yes
PSTN Caller Default DP: 1 Off Hook While Calling VoIP: no
Line 1 Signal Hook Flash To PSTN: Disabled



Disconnect Tone: 400@-30,425@-30;1(.75/.75/1+2)


Dial Plans Dial Plan 1:



here above i mentioned the configuration settings. if i did any wrong, please clarify me.

Thanks & Regards,
quest

[hwittenb]

The VoIP-To-PSTN Gateway is what is used for this type of call.

Using direct ip dialing you need to include the port number of the pstn line tab. In other words, if the default port number of 5061 is used, the sip uri used in SJ Phone would be ip_address:5061 where ip_address is the ip address and 5061 is the port number. If you have any entry in the userid field you would need to also include that in the address (userid@ip_address:5061).

On the pstn line tab you need to have register: No, Ans Call Without Reg: Yes

On the PSTN Line tab you would have Line Enable: YES

There is also another way to configure the SPA3102 using the Line 1 tab and call forwarding on the User 1 tab.

[maxipayne]

Hi
This is maxi,
The SPA3000 is a small unit (which is made of plastic and very light) and is designed to provide VoIP (Voice over IP) capabilities by interfacing with a normal analogue telephone (or cordless) and a standard PSTN line.

In technical terms, this has both an FXS and an FXO interface - the FXS interface allows a normal telephone to be turned into an IP phone and the FXO interface provides connectivity to a PSTN line (or of course another voip adapter which is locked by the provider). These interfaces can be configured independantly using the onboard web interface where when you log in as an admin user and switch to advanced mode, there are hundreds of settings to configure every possible value or paramter possible on each port.
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maxi

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