help with SPA-3102 simple voip gateway setup

[dnordenberg1]

Hi

I can't get my SPA-3102 to act as a simple voip gateway. I want to use my gigaset c610ip to dial out to the PSTN network. I always get "registration at provider failed" on the gigaset no mather what I try on the SPA-3102. If I enable Make Call Without Reg I can get a dialtone on the gigaset but this is as far as I made it yet. Incoming calls on PSTN should also be forwarded to the gigaset. Simply I want to use ethernet as an telephone extender on my lan. What settings do I have to make from SPA-3102 factory defaults to make it work? This is all on my lan so STUN or proxies should not be needed?

Kind regards

David

[Howard Wittenberg]

The SPA3102 is a sip client, not a server.  It does not have the ability to respond to sip Register requests from another sip client.  Your ip phone would need to have the ability to make peer to peer sip uri calls without registration to directly use the SPA3102.

I am not familiar with the gigaset c610ip.  I did glance thru the User manual.

If the phone needs to register to make calls then there are 3d party alternatives where you can register to the 3d party server and it will make the call to the SPA3102.  PBXes.com has a service they call SubPBX that will do this.  Sip2Sip.info is a service, and of course you can setup your own asterisk server.

The SPA3102 PSTN Line Tab needs to receive an incoming sip invite in the proper format to bridge the call out the attached PSTN Line.  Consumer products often will not make calls without registration to a distant server.  Some products do have options to make calls without Registration and if that is the case can often be make to work satisfactorily.  Troubleshooting is greatly enhanced if you have some way to capture (sniff) the ethernet packets.

If the ip phone has a setting to not register a configuration and a setting to make call without registration then I would setup a configuration with the proxy as the SPA3102 ip address:5061 where 5061 is the configured sip port number on the SPA3102 PSTN Line Tab.  You could initially setup the userid and password as anything to get it working without security.  When you dial a number for a call it would be sent to the SPA3102 to dial on the FXO port.

The VoIP-To-PSTN Gateway would need to be enabled, initially I would set the VoIP Caller Auth Method to none, One Stage Dialing should be yes, and the default dial plan for the Voip-to-PSTN Gateway would be (xx.).

If you are on a local network you usually would not have to forward ports in the router, if it was on a different network you definitely would have to forward ports in the router to the SPA3102 or the router would block the packets.

For calls from the PSTN to the ip phone I would start by configuring the dial plan of the PSTN-to-VoIP Gateway to automatically make a sip uri call to userid@ip_address:port of the ip phone .... (S0<:userid@ip_address:port>)

On the SPA3102 Line 1 Tab set Enable IP Dialing: Yes.  The incoming call will go to the PSTN-to-VoIP Gateway after the expiration of the PSTN Answer Delay setting.  I would set the Answer Delay to zero to start with.

The ip phone would have to accept unregistered incoming calls.

[dnordenberg1]

Hi!

Thanks for the long answer Ah, I think I understand a little more of how sip works now. I did not understand the separation between sip clients and sip servers and that peer to peer is possible without a server. Gigaset does not have a unregistred mode but as I wrote first, I can dial out if I allow unregistred calls on the SPA3102. So I guess I need UDP port 5060 open on the gigaset (which all SIP clients has?) and for it to accept an unregistred incoming connection on that port which it might not do?

I will experiment a litte more with this later today

Kind regards

David

[dnordenberg1]

Hi!

Your information was very helpful. I experimented some more and it seems the gigaset can receive unregistred calls. But I cant get this to work correct anyway

When I dial out from the gigaset to PSTN I hear a a dialtone for a second and then it goes quiet. No ringtones either but the other end is ringing and can answer and hear the gigaset but the gigaset does not hear the dialed end. But sometimes when I try to dial out it seems the gigaset gets a call from itself as soon as I hangup instead of calling out on the SPA-3102 PSTN (a loop in the dialplan?)

And I can sometimes I can dial in from the PSTN and even caller id works on the gigaset but this has only happend two or three times and most times it only rings forever without anyting happens on the gigaset (more evidence of a loop?)

I have attached my config as a html file.

Kind regards

David

[Howard Wittenberg]

When I dial out from the gigaset to PSTN I hear a a dialtone for a second and then it goes quiet. No ringtones either but the other end is ringing and can answer and hear the gigaset but the gigaset does not hear the dialed end. But sometimes when I try to dial out it seems the gigaset gets a call from itself as soon as I hangup instead of calling out on the SPA-3102 PSTN (a loop in the dialplan?)

The first problem you describe is commonly called one-way audio.  It has to do with one party or the other not sending the correct addresses.  Perhaps the gigaset is sending an external ip address instead of the local network address inside the sip INVITE.  I would attempt to pin down the problem by running sip debug traces on the SPA3102.

The "call from itself" probably has to do with the sip signalling regarding the disconnect.  The gigaset probably thinks you put the call on hold and it notifying you that the call is still connected as far as its concerned.

The SPA3102 picks up the caller id from the incoming pstn line call during the PSTN Answer Delay period.  In the U.S. it typically comes between the first and second ring and the PSTN Answer Delay needs to be 3 to 4 seconds to receive it.  There is a setting under the PSTN-to-VoIP Gateway as to whether or not you want to send the incoming caller id with the outgoing call.

Your configuration looks OK.  Thank you for including it.  That is helpful.  The VoIP Answer Delay is set to 3 seconds (instead of zero) which doesn't hurt anything, except you hear more rings, but you didn't hear the rings so I think the SPA3102 response to the Gigaset is not directed to the correct address.  The PSTN Answer Delay should probably be increased if you want to pickup the incoming caller id.

To run the sip debug trace you download and install a Syslog Program on your computer and put your local computer's address under Debug Server on the System Tab.  I see you already have an address under Syslog Server.  I would remove that because it just causes double entries.  On the System Tab set the Debug Level to 3.  On the PSTN Line Tab set the Sip Debug Option to FULL.  If you don't have a Syslog Program you can get a simple DOS one here:

https://supportforums.cisco.com/docs/DOC-9862

For the incoming call to the SPA3102 the trace is going to show you the detail of the incoming Sip INVITE and the SPA3102 responses and where they are being sent.  For the incoming PSTN Line call the trace will show if you pickup the incoming Caller ID, and show the detail of the outging Sip INVITE and any response received from the IP Phone.