SPA-3102 Transferring incoming voip calls to softphone

[psyafter]

There is a way to transfer incoming calls by voip provider (registered in Lin 1 tab) to softphone in same network?

I have connected my softphone to PSTN (with sipaccount created in PSTN line tab and proper configurations for pstn-to-voip and voip-to-pstn) and everything work fine (in/out calls on softphone).

Now I want transfer voip calls from my voip provider so I can talk on same softphone as well.

[Howard Wittenberg]

If I understand you correctly you are asking if you could switch an incoming call to an existing softphone instead of using the phone attached to the SPA3102.

You don't go into much detail on the current softphone accessibility except to say that the softphone can communicate with the SPA3102 PSTN Line Tab and you say it uses the same network. I do not believe the SPA3102 was designed to internally allow routing a call from the Line 1 Tab to the PSTN Line Tab voip, only to the PSTN Tab FXO port.

If the softphone is registered to a voip server you may be able to do an attended transfer of the incoming call directly to the softphone.

If the softphone is not registered to a voip server, then if you want to try something that is non-standard, since the softphone is communicating with the PSTN Line Tab you may be able to separately communicate with it using its local network ip address and port number. This, of course, will depend on the softphone capabilities.

I would first see if I could dial the softphone from the phone attached to the SPA3102 using a speed dial. In other words setup one of the Speed Dial Settings on the User 1 Tab with the softphone ip_address:port_number. In other words, 1777123123@168.1.104:7901 where 1777123123 is the softphone userid, 192.168.1.104 is the local network ip address and 7901 is the softphone port number. You may have to do some detective work to figure out the port number which hopefully doesn't change over time. Then see if you could dial the speed dial number from the phone attached to the SPA3102. If this is successful then you may be able to do an attended transfer of the incoming call by putting the incoming call on hold with a flash and dialing the speed dial number when the second dial tone comes up.

I roughly checked this out on my SPA3102 with an incoming call switching it to an X-Lite softphone on my computer. Depending on other settings, there may be some audio problems related to NAT settings on the SPA3102 that you will need to deal with.

[psyafter]

Hi. I mean auto forward (not manual transfer existing call) incoming call (from voip provider registered on line 1) to softphone by ip and url. I tried to forwad it on User tab (Cfwd All Dest) but no result, and in this case no information about voip call on Info tab (when no Cfwd All Dest so I see info about call).

I wand to work with softphone only without regular phone at all.

Now when I connecting phone to spa - I hear reorder signal. If I remove registration to provider (line 1) so regular dial tone is here. Something strange. When registering - incoming voip calls are ok and phone is ringing and I can talk with caller. But in this case no dial tone when hook off the phone...

[Howard Wittenberg]

Then you need to setup your softphone to register to an account at your voip provider and route incoming calls to your DID directly to your softphone.

You can either setup a 2d account or subaccount for the softphone or setup the SPA3102 to

Register: No.

I would setup Line 1 of the SPA3102 to not register (Register No) and to Make and Receive Calls without Reg. Many/most voip providers will allow you to make calls without registration.

The Cfwd All on the User Tab doesn't work for you because the SPA3102 itself doesn't do the forwarding, the SPA3102 sends a Sip command to your voip provider that there is a change in destination and it is up to the voip provider to send the call to the new destination. Most voip providers will not honor that request due, I believe, to fraud considerations. An Asterisk PBX and other Sip clients generally will honor the request.

The no dialtone is probably due to your setting of Make Calls Without Reg: No. In this case with this setting the SPA3102 does not give a dial tone when the SPA3102  is not registered to the voip provider.

[psyafter]

Thank you for answers.

I have voip phone that support only 1 sip account, so I wanted to forward pstn line and voip calls from voip provider to one device (currently I'm testing with softphone x-lite that support 1 sip account).

I can answer and call pstn line, but not voip provider calls.

Now I understand that it impossible with Line 1 tab.

But what about Pstn Line tab? There is option to register another Voip provider. What propose of second registration? Can I register my voip provider at this tab and it will work for in and out calls? May be so incoming voip call will be forwarded to softphone ip like with regular pstn calls?

//I tried one time to register but without success. I not really know what second sip registration do..

[Howard Wittenberg]

The SPA3102 PSTN Line Tab is designed to allow interfacing a legacy PSTN telephone circuit with voice over internet circuit. The PSTN Line Tab is more or less a separate voip adapter inside the SPA3102. It allows a PSTN caller to send his call to its destination over voip. Conversely it allows a voip caller to terminate his call on the public switched telephone network.

VoIP registration is a process that tells a voip server, a voip provider or PBX, the ip address and port number of a sip client that can receive a phone call and keep that server updated if the address changes.

You can setup a voip account on the PSTN Line Tab and register to a voip server and send incoming calls to the voip-to-pstn gateway or receive calls from the pstn-to-voip gateway and send them to a voip server for termination.

With the PSTN Line Tab incoming voip calls can only be routed to the voip-to-pstn gateway FXO port. There is no provision to allow the call to go out to another voip client like a softphone.

My impression of what you are doing with the softphone and the SPA3102 PSTN Line Tab is not the standard use of the SPA3102 PSTN Line capabilities. I believe you are using the PSTN Line Tab to allow a computer softphone to make and receive phone calls using your PSTN Line. I believe you are interfacing the SPA3102 with the softphone using direct ip addressing.

I would look for a softphone that allows you to configure multiple accounts. Setup one account for the direct ip calling and another to directly register with your voip provider to receive the incoming calls from your DID phone number. I am confident that you can find something to meet your needs.

[psyafter]

Hi Howard.

I tested voip provider at PSTN Line tab and used same dial plan (S0<:user@ip:port>) in PSTN Caller Default DP (this plan I use to forward incoming PSTN calls to softphone) and now incoming voip calls routed to softphone (!) after some rings on attached phone, but there is a problem: sound is poor and a lot of noise, line dropped after 3 seconds

[bruno.sdo]

I have a similar problem.

I would like to transfer incoming calls by PSTN to softphone (android) in same network?

What settings do I have to make?

[Dan Lukes]

Did you read previous comment in this thread ? Your question seems to be answered already. Several years ago.

[bruno.sdo]

Yes, but maybe I didn't understand it very well (because of the little knowledge of English) ...

I forgot to mention something ...
Can I forward a call received by the PSTN to a softphone, without using a SIP Server ?

[Dan Lukes]

SPA3102 is old and unsupported device. I have no one to test the configuration.  This message above claims it's possible and describe the method. The rest is on your own ...

[Howard Wittenberg]

Bruno,
You can configure the SPA3102 PSTN Line Tab dial plan to forward an incoming PSTN line call to a sip uri. As you know a sip uri is userid@ip_address:port_number. I believe you need to have the SPA3102 PSTN Line Tab settings configured with Register: No Make Call Without Reg: Yes, at least that was my setting when I tested it again today.  Edit:  I did run a satisfactory test where I forwarded an incoming PSTN call to a softphone installed on my pc which was on the same local network with the SPA3102.  I used WireShark logging to discover the incoming port number for the softphone, another alternative is to possibly get it from your voip provider.

A Sip URI PSTN Line Tab dial plan entry would look like (S0<:user_id@ip_address:port) If you were sending to port 5060 you would not need to include the port number. Of course you would code the PSTN Caller Default DP to that dial plan number.

I believe you are going to have trouble sending a sip uri directly to a softphone for a number of reasons
1. Softphones are designed to register to a sip server and may not allow incoming calls from unsolicited senders, or may not even work if not registered.
2. Softphones use a wide number of different sip port numbers (for incoming calls) that are difficult to discover, may change over time, and are required in a sip uri. A softphone may have a setting for an incoming sip port number, but that is not common.

Some voip providers have a scheme to allow incoming sip uri calls and/or forwarding outgoing sip uri calls. It is much easier to use one of these features for one leg of the call or to just use a voip provider without using a direct sip call.