connect soft phone from internet to spa-3102 to dial on pstn FXO port

[noelgallagher85]

Hello:-)

My question is very simple. I've already read a lot of documents and previous posts about this operation.

As you can read from title i would like to install on my Android phone a softphone which could connect from internet (not in internal lan) to spa-3102 (when of course already setted fowarding ports, sip rules, static public ip address) for dialiang a number with my home PSTN line where i've got an active plan for calling. In this way i could use my home line from my cellular when i am out.

I've read that this should be possible setting call "without registration" and so i'm really interested in this product but:

0) Can i really do this?

1) how safe is doing it? how can i be sure that nobody else, except for me, will use my line (if there is not any kind of registration)? What about protection?

2) do you know any softphone for Android platform that can be used with option calling "without registration"?

Thank You Francesco

[Howard Wittenberg · Accepted solution]

0) Can i really do this?

It is possible, although it can be a challenge to get it working.  The hardest part is setting up the Android phone.

1) how safe is doing it? how can i be sure that nobody else, except for me, will use my line (if there is not any kind of registration)? What about protection?

You can setup the SPA3102 to ask for a user id and an encrypted password for the call which gives you protection.  You do this with "http digest" authentication.

2) do you know any softphone for Android platform that can be used with option calling "without registration"?

In "expert" mode you can setup the "CSipSimple" android softphone to dial without registration.  The "expert mode" secret is outlined here:  http://code.google.com/p/csipsimple/issues/detail?id=981

My notes say to "Do basic wizard first, then switch to expert, then remove registrar url".

You setup your SPA3102 ip address (symbolic or static) along with the PSTN Line sip port number as the CSipSimple "Proxy URI".  I would change the sip port number on the SPA3102 PSTN Line Tab to something other than 5061 for additional security.   

You setup the SPA3102 PSTN Line Tab for http_digest authentication which means it will challenge the incoming call for a username and password which you have setup on your CSipSimple softphone.  This solves your authentication problem and security problem you mentioned.  This sets up the SPA3102 for one-stage dialing out the FXO (Line) port.

As you know on the SPA3102 PSTN Line Tab under VoIP-To-PSTN Gateway you select  VoIP Caller Auth Method: HTTP Digest.  You put your username under VoIP User 1 Auth ID: and your password under VoIP User 1 Password.

The userid and password you setup on the SPA3102 as described here is the same userid and password you setup under CSipSimple on the softphone.

With direct ip address inbound calling to the SPA3102 you MUST forward your PSTN Line Sip Port Number in your router to the SPA3102.  This is a must do because the incoming call is not solicited and if not forwarded to the SPA3102 your router's firewall will discard the incoming call packet.

For this response, I ran a test over my local network with my SPA3102 and CSipSimple and it works.  Over the internet shouldn't be any different except you may have to troubleshoot some external ip address network problems.

[noelgallagher85]

What should i tell to you?

Just thank you could not be enough!

This was exactly what i'm looking for and you answered all my questions! ALL! You are amazing!

THANK YOU!

I'm going to buy a SPA3102!!!

THANK YOUUUUU!!

[BobFall]

Hi. I am trying to configure an SPA3102 with a CSipSimple in a similar scenario as above but with little success.  I can't get the basic operation going at all and wonder if I could get more detailed instructions please.

My requirement is as follows:

Make calls to the PSTN line from CSipSimple on an Android device, through a VPN to the local network where SPA3102 exists.  VPN is handled by the Android device and is fine.

SPA3102 not to answer any PSTN incoming calls automatically, but just route the incoming call to the CSipSimple.  If VPN is not connected or CSipSimple not running, then incoming call to ring indefinitely until answered by the PSTN's own voicemail.

Password required by SPA3102, and configured in CSipSimple for making/receiving PSTN calls.

The above may seem simple enough but I can't get it going.  The SPA3102 is on static IP, its WAN port is connected to the LAN switch, and its Line port is connected to the PSTN master socket.  The 'Expert' setup in CSipSimple has too many fields to fill.

Thanks

[BobFall]

To take a step-by-step approach, I'd like to start by keeping all devices on the same LAN to reduce router/firewall/port forwarding/static IP issues getting on the way.

I undrestand that the SPA cannot accept registration from a SIP client, but CSip can be configured to make calls without registration, in Expert mode.

As a basic setup, I have configured the SPA on static IP on the LAN. The WAN port is connected to the router/gateway of the local network, and the LINE port is connected to my PSTN line.

From the default state, I have performed the following settings on the "PSTN Line" tab of SPA3102:

Line Enable: YES

SIP Port: 5070

VoIP-To-PSTN Gateway Enable: YES

PSTN-To-VoIP Gateway Enable: YES

On CSipSimple: I have configured a Basic account with dummy UID and Password, and the Server address being: static_IP_of_SPA:5070

I then converted the account to Expert mode and cleared the field marked 'Registration URL'. As per the instructions above to allow calling without registration.

Issues: CSip says Registered but when I dial a number to call out, it makes an attempt through to the SPA but quickly disconnects and drops back to the dialing pad. No error messages.

If I make a call to the PSTN line using another mobile, after around 8 seconds the SPA answers the call and plays a dialing tone to the caller as if it expects the caller to dial a number, however for only around 5 seconds before the SPA terminates the call.

I am obviously not even close to the desired goal, and would like to ask if anyone can please lend a hand in getting this going.

Many thanks in advance.

[Howard Wittenberg]

I think you are pretty close.  When it doesn’t work I would be running a sip trace on the SPA3102 to get clues as to what is going wrong.  To run a sip trace you download and install a syslog program on your pc, you put your pc’s local network address under Debug Server on the System Tab, Set the Debug Level to 3, and on the Line 1 Tab and the PSTN Line Tabs you set the Sip Debug Option to Full. I believe you can download a simple syslog program here.

In your testing, I would separate the tasks to try to isolate the problems.  Get something working and then move on from there.  First I would just get the softphone to place a call to the SPA3102 Line 1 hardware attached phone.  This would just require addressing the call to the userid@ip_address:port on Line 1 of the SPA, port forwarding if the calling softphone is not in the local network, and setting the Line 1 configuration to receive calls without registration. 

The above test should isolate most network problems.  With this test you should have the phone ringing and a two-way talk path. 

After the simpler application is working I would change the called sip url to call the SPA3102 PSTN Line.  You could first get it calling the userid@ip_address:port and get a dial tone.  Then change it to called_pstn_number@ip_address:port for one stage dialing to access the SPA3102 voip-to-pstn gateway and automatically dial the number on the pstn line.  Do it first without security and after that is working add the password security.

After incoming voip calls to the SPA3102 are working, then setup the SPA3102 PSTN to VOIP gateway to automatically call the distant softphone on an incoming call from the PSTN line.  Hopefully the softphone ip address is static or symbolic or you could have later problems.

Dialing is a little tricky on CSipSimple.  Dial the full sip url with userid, ip_address and port number.  I would be using non standard port numbers for the Line 1 and the PSTN Line Tabs.

As you know, your described use of the SPA3102 and android softphone is very specialized and not very common.   Except for the one stage dialing The application could be solved in a more conventional way using your own inexpensive PBX server or a conventional commercial voip sip server service.

One thing to check, your PSTN Line needs to have a voltage level when on hook above the Line in Use Voltage setting which defaults to 30v.  With a regular PSTN line that is usually about 48v.  Otherwise the SPA will think the line is busy.