Phone to SPA3102 to 3CX to SIP Phone

[brianmcdaniel]

All,

I'm trying to connect an analog phone to the SPA3102 to a P.C. running 3cx to a SIP phone also connected to the P.C. via a switch all locally through switch so no gateways or anything else to connnect to anything outside at the moment just a local network on 192.168.0.xxx.
I have the Internet port of the SPA3102 connected to the local network and the Ethernet port of the SPA3102 is connected to a laptop for configuring the device (192.168.1.3) and the phone port of the SPA3102 is connected to an analog phone.
Currently I'm able to talk to the SIP phone via the 3cx VoIP Client, but I'm unable to talk to the analog phone through the VoIP Client (I get the "Destination Not Found" error). The analog phone also has no dial-tone. I've gone through countless different configurations but to no avail as of this moment the extension for the Line (5000) shows that it is registered and I have the following settings:

P.C. running 3CX IP 192.168.0.102
VoIP Client is extension 1000.
On 3CX I have A SPA3102 with an extension and user id and passwords etc.. set to 5000 with the IP set to 192.168.0.88 port 5060.

Status for SPA3102 IS:

Product Information
Product Name: SPA-3102 Serial Number: FM600F200144
Software Version: 5.1.5(GWa) Hardware Version: 1.1.5
MAC Address: 000E08CB625E Client Certificate: Installed
Customization: Open

System Status
Current Time: 1/1/2003 12:14:34 Elapsed Time: 00:14:34
Wan Connection Type: Static IP Current IP: 192.168.0.88
Host Name: SipuraSPA Domain:
Current Netmask: 255.255.255.0 Current Gateway: 0.0.0.0
Primary DNS:
Secondary DNS:
LAN IP Address: 192.168.1.3 Broadcast Pkts Sent: 0
Broadcast Bytes Sent: 0 Broadcast Pkts Recv: 1201
Broadcast Bytes Recv: 81742 Broadcast Pkts Dropped: 0

WAN IP is set to: 192.168.0.88
The Proxy for line is 192.168.0.102
SIP port 5060.

Any help would be greatly appreciated and if someone needs more information about the setup I'll be more than happy to provide it.

Brian McDaniel
Senior Software Engineer
Teradon Industries

 

[brianmcdaniel]

UPDATE:

I'm able to call the analog phone via one of the SIP Phones or the 3CX VoIP client, haven't figured out how to call from the analog phone to the VoIP client or an SIP phone. Also once I get this figured out I will need set up the PSTN connection on the SPA3102 to talk to a custom VOX card that we have.

 

[Henk]

UPDATE:

I'm able to call the analog phone via one of the SIP Phones or the 3CX VoIP client, haven't figured out how to call from the analog phone to the VoIP client or an SIP phone. Also once I get this figured out I will need set up the PSTN connection on the SPA3102 to talk to a custom VOX card that we have.

To use the analogue phone to dial a SIP phone or a 3cx VoIP client you must dial the VoIP number that was provided to you by the VSP (DID number). You cannot dial extensions or SIP addresses with an analogue phone (FXO port).

H.

 

[brianmcdaniel]

Thanks for the reply.
For clarification - I'm not really using the 3102 to connect to VoIP service I'm using it locally, i.e. within a private data and phone network.

Through the local network I'm able to dial my SIP phones from the Analog phone via the 3120, I just had to add the SIP phones I wanted to talk to the dial plan and define the analog phone as a SIP extension within 3CX (if you define the analog phone using as a PSTN device it doesn't work). So I'm able to talk to my SIP phones and my analog phone, also I'm able to connect the 3102 to our local phone network and extensions within our network are able to dial in to the 3102 which are routed to a the VoIP Client on the the P.C. The last piece I can't get working is dialing out of the PSTN from either the SIP phones or the VoIP client to an extension on our phone network, I've defined a default rule so that calls starting with 0 go to the PSTN, but I don't get a dial-tone like I expect, almost there though

Any ideas?

 

[Henk]

Thanks for the reply.
For clarification - I'm not really using the 3102 to connect to VoIP service I'm using it locally, i.e. within a private data and phone network.

You either do or you don't, so I assume you do not have the VoIP service configured in the SPA (eg login details)

The last piece I can't get working is dialing out of the PSTN from either the SIP phones or the VoIP client to an extension on our phone network, I've defined a default rule so that calls starting with 0 go to the PSTN, but I don't get a dial-tone like I expect, almost there though

Any ideas?

Not sure if I understand this.

You dial from the PSTN in? or do you want to use your sip phones to dial out via the PSTN?

Dial tone is provided either from PSNT line, your VSP or 3CX.

H.

 

[brianmcdaniel]

Thanks for the reply.


That's correct I don't have the VoIP service configured in the SPA - I'm not using any VoIP service at this time.

I'm able to dial in (in being the 3cx server) from a local telephone network via the PSTN line on the SPA. I'm not yet able to dial out of the same line via a SIP phone to a local extension on the local network - the dial tone is provided by the local network.

 

[brianmcdaniel]

Still unable to get things working properly, this really shouldn't be that difficult :x

If I unplug the spa3120 I'm able to call from the analog phone to an extension on the local PSTN so I know the PSTN is good.
However, once I plug the spa3120 I'm no longer able to call an extension on the PSTN.
I've included <#9:>xx.<gw0> in the dial plan for the Line but when I press #9 I don't hear a dial tone.

 

[N_G]

There is always the option of buying a commercial edition with technical support...... From reviewing these posts i think this would help you.