SPA-3102 - Sipgate + VoIpDiscount

[onlineuser]

Hi,

I have connected the SPA-3102 to extension 2 of my telephone-system.

Now when I dial the extension from the SPA-3102, I can make phonecalls via SIP.

Now I want to use sipgate (inbound and outbound) and voipdiscount (only outbound) on my SPA-3102. But all other analogue extension should be able to make phonecalls via SPA.

So, when any extension rings the SPA, it picks up the call and I hear the dialtone from SIP.

How is it possible that the user can choose between sipgate and voipdiscount for outgoing calls?

When a inbound-call via sipgate arrives the SPA the call should be forwarded to extension 3 on the pstn-line from the telephone-system. How can SPA-3102 manage this configuration?

Regards

[Howard Wittenberg]

If I understand correctly your "telephone-system" is analog and "extension 2" is attached to the Line (FXO) port of the SPA3102.

There are different ways to configure the SPA3102 for the task you outline.  One way to configure it would be to setup your SipGate account on the PSTN Line Tab for incoming and outgoing calls, and setup VoIPDiscount as a Gateway Account under Gateway Accounts on the Line 1 Tab.  I would assume you wouldn't use the phone (if any) attached to the SPA3102.

How is it possible that the user can choose between sipgate and voipdiscount for outgoing calls?

You would do this with the dial plan, the PSTN Caller Default DP mentioned below.

When a inbound-call via sipgate arrives the SPA the call should be forwarded to extension 3 on the pstn-line from the telephone-system. How can SPA-3102 manage this configuration?

You also do this with the dial plan, the VoIP Caller Default DP mentioned below

Setup your Sipgate account on the PSTN Line Tab (Proxy/Userid/Password) and Register: Yes. and probably NAT Mapping Enable: Yes.  You may also want to set Make Call Without Reg: Yes

You could setup an outgoing dial plan with #9 to precede calls to be sent thru VoIPDiscount, other calls thru SipGate with this dial plan.  You don't have to use #9, you could use any prefix you find convenient. 

Setup PSTN Caller Default DP: 1

Setup Dial Plan 1 as (<#9,:>xx.<:@gw1>|xx.)

Setup the PSTN-To-VoIP Gateway Setup with

PSTN-To-VoIP Gateway Enable: yes

PSTN Caller Auth Method: none

PSTN Ring Thru Line 1: no

PSTN CID For VoIP CID: no

PSSTN Caller Default DP: 1

Setup VoIPDiscount on the Line 1 Tab under Gateway Accounts as Gateway 1

Gateway 1: userid@sip.voipdiscount.com

GW1 Auth ID: userid

GW1 NAT Mapping Enable: yes

GW1 Password: voipdiscount_password

For incoming voip calls, you would manage sending an incoming SipGate call to extension 3 thru the following dial plan

Setup VoIP Caller Default DP: 2

Setup Dial Plan 2:

(S0<:3>)

(assuming 3 is how you call extension 3)

You would have the rest of the VoIP-to-PSTN Gateway setup with:

VoIP-To-PSTN Gateway Enable: Yes

VoIP Caller Auth Method: none

One Stage Dialing: Yes

VoIP Caller Default DP: 2

Other settings on the PSTN Line Tab would include

VoIP Answer Delay: 0

PSTN Answer Delay: 0 (unless you care about capturing an incoming caller id on the PSTN Line)

[onlineuser]

Thanks for detailed answers!

Yes, I have a analog telephone system, on extension 2 is SPA-3102 on FXO port. Now I do not use the line-port on SPA-3102 but if it's possible I also want to use it for incoming sipgate and outgoing calls via SIP (not for calls over the analog telephone-system).

I will try your settings! :-)

Is it also possible to set the voipdiscount account to PSTN - because outgoing calls via the SPA-3102 only goes via voipdiscount, the sipgate account is only for incoming calls - so, when any extension from the analog system calls the SPA-3102 the voipdiscount-dialtone should be first.

Is there a HOWTO how the dial-plans work?

On my TA612V I could choose the local SIP port so the sipgate server always connected to my firewall on port 5083.

The SPA-3102 and a lot of other SIP router doesn't have this option any longer.

I tried to set the EXT Sip Port on SPA-3102 to port 5084 but the sipgate server always used other port for connecting to me (the port also changed every 5 or 6 minutes). I don't use NAT on my DMZ where the SPA3102 is connected.

Is there a way to reach that sipgate always connects on the same port to me, like my TA612V?

Even when I connect to sipgate via sipgate proxy the sipgate server tries to connect to my public IP on different ports instead of via the existing proxy-connection.

Regards

[Howard Wittenberg]

Now I do not use the line-port on SPA-3102 but if it's possible I also want to use it for incoming sipgate and outgoing calls via SIP (not for calls over the analog telephone-system).

The FXO port is an analog interface it is not digital.  To use it for digital, sip protocol, calls you would have to interface it to an FXS port of another analog telephone adapter.

Is it also possible to set the voipdiscount account to PSTN - because outgoing calls via the SPA-3102 only goes via voipdiscount, the sipgate account is only for incoming calls - so, when any extension from the analog system calls the SPA-3102 the voipdiscount-dialtone should be first.

When an extension from the analog system calls the SPA-3102 the SPA-3102 provides the dial tone and the dial plan logic in the SPA3102 determines the outbound call routing.  In my dial plan example, if you want the default outbound calls to go to VoIPDiscount and have to dial a special prefix for SipGate you would reverse the settings .... (<#9,:>xx.|xx.<:@gw1>)

Is there a HOWTO how the dial-plans work?

See the Cisco ATA Administration Guide

http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf

There are also some threads in other 3d party forums.

Is there a way to reach that sipgate always connects on the same port to me,

The SPA3102 Line Tab and the SPA3102 PSTN Line Tab each have SIP Port: setting.  You can set these SIP Ports to any number you want.  In your case set the Sip Port on the PSTN Line Tab to 5084.  This is the port used for incoming communication with the SPA3102.

[onlineuser]

Thanks for your answers! :-)           

Now I do not use the line-port on SPA-3102 but if it's possible I also want to use it for incoming sipgate and outgoing calls via SIP (not for calls over the analog telephone-system).

The FXO port is an analog interface it is not digital.  To use it for digital, sip protocol, calls you would have to interface it to an FXS port of another analog telephone adapter.

Ok, but I meant that my telephone system (extension 2) is connected with SPA-3102 (line port - seems to be the FXO).

On the phone-port (seems to be FXS) from SPA-3102 there is at moment no phone connected.

But when I would connect a phone to it (SPA-3102 is in server room) - would following things possible?

- any extensions dials extension 2 and they hear SIP dialtone (works now)

- per default the SIP call goes via voipdiscount (works now)

- with extra dial plan SIP call goes via sipgate if user wants

- when a sipgate call comes in, the call will be routed back to analog phone system to extension 1 (maybe with special callid on display) (maybe also the phone on Line 1 from SPA-3102 can ring)

- when I connect a phone to the phone port of SPA, is it possible to make SIP calls via voipdiscount and sipgate (calls via extension 0 of analog phone system are not necessary) - moreover my analog phone system doesn't accept calls from extension 2 to extension 0 (POTS)

Is there a way to reach that sipgate always connects on the same port to me,

The SPA3102 Line Tab and the SPA3102 PSTN Line Tab each have SIP Port: setting.  You can set these SIP Ports to any number you want.  In your case set the Sip Port on the PSTN Line Tab to 5084.  This is the port used for incoming communication with the SPA3102.

I tried this - on sipgate website I saw that my device connects from 5084 but the sipgate server tries to establish a connection to me via other port.

On my TA612V I set my local port to 5083 - I also saw this on sipgate website and there sipgate always tries to connect to me via port 5083.

It seems to be a "strange" problem - I will disconnect my TA612V - maybe it's because I use both devices at same time.

Regards

[Howard Wittenberg]

On the phone-port (seems to be FXS) from SPA-3102 there is at moment no phone connected.

But when I would connect a phone to it (SPA-3102 is in server room) - would following things possible?

- when a sipgate call comes in, the call will be routed back to analog phone system to extension 1 (maybe with special callid on display) (maybe also the phone on Line 1 from SPA-3102 can ring)

If you have another SipGate registration (user?) and you setup that different registration on the Line 1 port then you can setup an incoming call to forward to your extension 1.  If you want the phone attached to the SPA3102 to ring you can set it up to ring first and if no answer then forward to extension 1.  You can't have both the phone attached to the SPA3102 and your extension 1 ring at the same time you need to ring the phone attached to the SPA3102 first and then if no answer ring the extension 1.  You need to register this different SipGate account on the Line 1 Tab and the forwarding is done on the User 1 tab.  On the User 1 Tab you would either forward all calls or forward on no answer to 1@gw0

- when I connect a phone to the phone port of SPA, is it possible to make SIP calls via voipdiscount and sipgate (calls via extension 0 of analog phone system are not necessary) - moreover my analog phone system doesn't accept calls from extension 2 to extension 0 (POTS)

Yes you can make calls from the phone attached to the SPA3102 out thru VoIPDiscount and out thru the 2d SipGate account that you have setup using a dial plan that you setup on the Line 1 Tab.  The dial plan would be similiar to the dial plan you setup on the PSTN Line Tab.

I tried this - on sipgate website I saw that my device connects from 5084 but the sipgate server tries to establish a connection to me via other port.

On my TA612V I set my local port to 5083 - I also saw this on sipgate website and there sipgate always tries to connect to me via port 5083.

It seems to be a "strange" problem - I will disconnect my TA612V - maybe it's because I use both devices at same time.

The problem would certainly be that you are registering from both TA612V and the SPA3102 ... two separate addresses (sip ports). In this case usually the last one to register will be the address where they send the call.  I am not knowlegable about the Netgear TA612V router and voice adapter. The SPA3102 has a router function that you can use.  I do not believe I would use the TA612V unless you know how to disable its voice functions.

[onlineuser]

Ok thanks! I will try it to configure! :-)

[onlineuser]

one more question:

Does it matter if I first enter the sipgate account at PTSN-tab and then the voipdiscount-data under gateway 1 or can I also enter first the sipgate-data and then the voipdiscount-data?

I think it should be equal because the dial-plans are deciding which account should be used?

[Howard Wittenberg]

It doesn't matter which way you do it.

I should point out again that the SipGate registration you enter on the Line 1 Tab is different than the SipGate registration you enter on the PSTN Line Tab.  The "User ID" would be different.  This is because you said you wanted different routing for different incoming calls.

[onlineuser]

Ok, thanks.

For Line1-tab I will use the same voipdiscount-account as for PTSN-tab.

I don't need the sipgate-account on Line1, because the simultaneously ringing on PTSN-sipgate and Line1-sipgate won't work for incoming calls.

And the phone on Line1 I only need for make calls from server-room (I hope that voipdiscount allows 2 phonecalls at same time - this happens when anyone calls via analog-phone-system via PTSN and when I make a call via Line1).

[Howard Wittenberg]

If you don't have SipGate setup to register on Line 1 then you need to set Make Call Without Reg: Yes on Line 1.

[onlineuser]

I have a problem :-)

Now I configured my PSTN with my sipgate account. This works fine.

Because on PSTN is no way to configure my voipdiscount account for outoging calls I followed your tips and I have activated the Line 1. First I tried to fill the voipdiscount data to SIP setting on Line 1 and then only in Gateway Accounts on Line 1 (SIP fields stayed empty).

Making calls from analogue telephone system via sipgate works fine, also incoming sipgate calls on PSTN will be routed to only phone #3. :-)

But when I call extension 2 from any phone I hear the PSTN dialtone - but it's not possible to make a call via voipdiscount which is configured on Line 1.

When I enter voipdiscount data at SIP Account on Line 1 the SPA-3102 can register!

Do you have any idea which I could try?

[Howard Wittenberg]

I'm not sure I understand your problem.  It is true it is not possible to make a call via voipdiscount which is configured on Line 1 as the primary provider but it is possible to make a call via voipdiscount if you have it setup properly under Gateway Accounts.

A bridged call from the incoming Line Port (FXO port) on the SPA3102 can access a voip configuration setup under "Gateway Account" on the Line 1 Tab.  It cannot access a configuration setup as the primary voip provider on the Line 1 Tab, it can only access the configuration setup under "Gateway Account" (or the configuration setup on the PSTN Line Tab).  You determine the routing via the PSTN Line dial plan as I previously posted.

You put the outgoing configuration for voipdiscount on the Line 1 Tab under Gateway Accounts in this format.  Note: it is not the same format that you use for the primary provider:

Gateway 1: your_voipdiscount_userid@sip.voipdiscount.com

GW1 Auth ID: your_voipdiscount_userid

GW1 Password: your voipdiscount password

GW1 NAT Mapping Enable: yes

A call from the analog phone attached to the SPA3102 can access the primary voip provider configured on the Line 1 Tab, it can also access the voip configuration setup under Gateway Accounts on the Line 1 Tab (@gw1, @gw2, etc), and it can also access analog Line Port (FXO port) (@gw0).  The call routing is determined by the dial plan you setup on the Line 1 Tab.

[onlineuser]

Ah ok, but when I enter the voipdiscount settings only as gateway, then on status site there Line 1 stays unregistered. So I also can disable Line 1 and the gateway-settings for outgoing calls should work fine.

But the dislplans are confusing.

I set the dialplan on PSTN that incoming calls from PSTN (sipgate) will be routed to analogue extension 3 - this seems to work fine.

But now, when any phone wants to call via voipdiscount I must set the dialplan in PSTN - at Line 1 I only can define one dialplans.

Moreover, it would be more logical if I set the dialplan for dialing via voipdiscount gateway in PSTN settings!?

Is this correct?

If this works, I could activate Line 1 with voipdiscount-settings to use a phone on Line 1 port for making outgoing calls? Or is it not possible to use a voipdiscount account for two calls at same time. If yes I could use Line 1 with any other voip-provider!?

[Howard Wittenberg]

when I enter the voipdiscount settings only as gateway, then on status site there Line 1 stays unregistered.

That is correct.

So I also can disable Line 1 and the gateway-settings for outgoing calls should work fine.

If you are using the gateway setting from the PSTN Line Tab you are not using the settings on the Line 1 Tab

But now, when any phone wants to call via voipdiscount I must set the dialplan in PSTN - at Line 1

No it is not correct.  For bridged outgoing calls from an incoming call on the Line Tab (FXO port) your dial plan on the PSTN Line Tab would reference the gateway.  For example I previously posted a PSTN Line Tab dial plan for outgoing voip calls:

(<#9,:>xx.<:@gw1>|xx.)

where you dial #9 plus your number and the call is sent to the gateway 1 configuration (voipdiscount)

If you do not preference your number with #9 and the call is sent to SipGate which you configured on the PSTN Line Tab.

The Line 1 Tab dial plan is only for calls initiated from the analog phone attached to the SPA3102.

I could activate Line 1 with voipdiscount-settings to use a phone on Line 1 port for making outgoing calls? Or is it not possible to use a voipdiscount account for two calls at same time. If yes I could use Line 1 with any other voip-provider!?

I do not know if voipdiscount would allow more than one simultaneous call at a time.  I doubt that they would.  You could setup another provider on the Line 1 configuration or on one of the three remaining gateway configurationsl..