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SPA3102 Call forward PSTN to VOIP

Level 1
Level 1

Is there a way to conifgure the 3102 to always forward or hot dial a voip number whe a pstn call comes in. Seems like it only works for pstn user(phone- fxs)

Al N

14 Replies 14

Level 1
Level 1

Yes... under PSTN Line tab:

PSTN-To-VoIP Gateway Enable:yes

PSTN Caller Auth Method:none

PSTN Caller Default DP:2

Dial Plan 2:(S0<>)

PSTN Answer Delay:0

i thought you need the pstn number in the dial plan section? can you please clearfiy.

i have one and i am not able to forward calls from the voip to the pstn or the other way around

No PSTN No. in DP required , since this intended to forward incoming PSTN call to a SIP No., you only need target VoIP no. in DP.

thanks for the reply

i just want to make sure i understand it correclty please.

in DP1 i will use the following config

(S0<718xxxxxxx) my sip number

the device will be placed in my remote office in Egypt ,my pstn number in Egypt is 424xxxx

i need to be able to do the following:

1-for someone in the U.S to call the 718 number and have someone answer the call to attached analog phone

2-for someone in the U.S to call the 718 number and be able to acces the pstn in egypt and dial mobile number for example 012xxx

3-for someone in Egypt using mobile number 012xx to call the pstn number 424xxxx and get the call routed to voip number 718xxx so can call let us say 347xxx in the U.S

thanks again for your help

Before you mess with DP, did you get the 718 number working with your SPA3102? i.e. you need to register with VSP who provide your 718 sip number in Line1 Tab.  If this works, someone call your 718 sip number will ring the analog phone attached to the SPA3102 which is your requirement #1

Your requirement #2 need to enable VoIP to PSTN gateway in PSTN Tab once #1 working.

the first part is currently working, i am able to dial the 718 and ring the attached analog phone. but i need help conifguring the voip to pstn part please.

Depend on how you want it works...

Call the 718 registered on Line 1 Tab, it will ring the analog handset attached to the SPA3102. In User 1 Tab,  you can setup the call to be forwarded on no answer (Cfwd No Ans Dest) to gw0 and (Cfwd No Ans Delay) to 15, then if nobody pickup the call, it will forward to the PSTN gw0 after (Cfwd No Ans Delay) defined time and the caller will receive a dial tone to make an outgoing call on the Egypt PSTN network.

As an alternative, you can also setup the calls to 718 number registered on Line 1 Tab to be unconditionally forward to PSTN. Set (Cfwd All Dest) to gw0 in User 1 Tab. In this way, all calls will not ring the analog handset attached to the SPA3102 and directly giving caller dial tone to make an outgoing call on the PSTN network.

thank you so much

it works but when i hang up the first call and try to make another call it's busy signal? would you know why?

You may need to figure out the Disconnect Tone parameter of the PSTN network and redefine them in Regional Tab and PSTN Line Tab. Most likely the SPA3102 doesn't recognize the disconnect tone, on some network, even if the parameter is correct, it failed to detect, I have requested a fix on this issue in the firmware for year and so far still waiting...  see my original post here

thank  you so much. it's working now

Hi, I want to forward all of my incoming pstn call to a voip number. I have successfully registered my Sip account; but now I cannot forward the call. My sip gateway call method is country code+phone no. Could you help me how to input this into SP3102? I tried by input in Dial Plan (SO<:6581090738) but doesn’t work. Can you help me? Many thanks in advance.

 I tried by input in Dial Plan (SO<:6581090738) but doesn’t work



For forwarding a PSTN Line call to an outgoing voip call the format of the Dial Plan entry should be S0 (S zero) not SO ... (S0<:6581090738>)


The forwarding assumes the following:
Dial Plan 2: (S0<:6581090738>)
PSTN Ring Thru Line 1: No
PSTN Caller Default DP: 2
PSTN Caller Auth Method: none
PSTN Answer Delay: 0 or 3 or 4 Depending on whether you want to pickup the incoming caller id
You have setup your voip account credentials on the PSTN Line Tab (Proxy, UserID, Password)
You have enabled the PSTN Line Tab and the PSTN-To-VoIP Gateway
You have configured NAT Mapping Enable correctly


If it doesn't work you should run a sip trace of a failed call and start a new thread instead of using this old thread.

Dear Sir,

Many thanks for your great input.
It works beautifully now.

Btw, to change forwarded no, you need an admin log in.
Is there any way to change forwarded no in S0<:xxxx from attached analog phone?
Any code to change xxxx above?

Thank you very much.

Best regards,
Deddy H.

Btw, to change forwarded no, you need an admin log in.
Is there any way to change forwarded no in S0<:xxxx from attached analog phone?
Any code to change xxxx above?


No, there is no way to change the dial plan or the default pstn gateway dial plan number without logging into the configuration.


You might be able to change a forwarding destination for specific caller id's using the User Tab. 


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