SPA-3102 Setup Help

[ess]

Hi,

First let me start by describing the current system.

3cx Server: 192.168.1.100
PSTN Gateway (Linksys 3102): 192.168.1.200 | Internal Number: 10700
Sip Phone 1 (Linksys SPA-921): 192.168.1.202 | ext. 500
Sip Phone 2 (Linksys SPA-921): 192.168.1.203 | ext. 501

I do not have any VoIP providers, only a PSTN line.

All devices show as "Registered", and all were chosen because guides were available. The configurations for each device are setup as recommended, I believe.

I am able to dial between extensions without a hitch. The problems start when I try to involve my gateway.

For outgoing calls, I receive a busy tone. I setup the recommended dialing plan from the 3cx guide for a 3102, and I have an outbound rule that numbers starting with 0 are masked to remove the 0 and then sent to the gateway. Doing this results in a busy tone, and the following status (I have removed my phone number):

11:27:12.013 CallConf::Rejected Call(C:31) is rejected: Not Found
11:27:12.013 StratInOut:nCancel Call(C:31): Call from Ext.501 to (Phone Number) terminated; reason: Not Found
11:27:11.662 CallConf:nIncoming Call(C:31): Incoming call from Ext.501 to sip:0(Phone Number)@192.168.1.100

I am not sure if this is because it is trying to use a voip provider or what.

For incoming calls, I am able to receive them according to my inbound rule pointing at extension 501! However, on the phone, each ring is shown as a call. In other words, if it rings 4 times, I have 4 missed calls from my gateway. The server reports the following:

11:31:31.706 StratInOut:nHangUp Call(C:34): Call from Ln:10700@Home to 501 has been terminated
11:31:27.029 CallConf:nProvisional Call(C:34): got response from 501
11:31:26.829 CallConf:nIncoming Call(C:34): Incoming call from Ln:10700@Home to sip:[email protected]
11:31:26.048 StratInOut:nHangUp Call(C:33): Call from Ln:10700@Home to 501 has been terminated
11:31:25.157 CallConf:nProvisional Call(C:33): got response from 501
11:31:24.876 CallConf:nIncoming Call(C:33): Incoming call from Ln:10700@Home to sip:[email protected]
11:31:24.045 StratInOut:nHangUp Call(C:32): Call from Ln:10700@Home to 501 has been terminated
11:31:20.200 CallConf:nProvisional Call(C:32): got response from 501
11:31:19.929 CallConf:nIncoming Call(C:32): Incoming call from Ln:10700@Home to sip:[email protected]

In addition, however probably unrelated, even though I have the answer delay set to "0" for pstn in the gateway configuration, the other standard telephones in the office ring, and the answering machine here picks up (probably because it is set for fewer rings than the voicemail configuration for the extension.)

Also, the server repeats the following error message throughout its operation:

11:34:58.624 Endpoint::findSource Can not find source by following specs: Contact IP: ''; From: '10700'; To: ''; Rline: ''; Contact: ''
11:34:43.512 Endpoint::findSource Can not find source by following specs: Contact IP: ''; From: '10700'; To: ''; Rline: ''; Contact: ''
11:34:28.410 Endpoint::findSource Can not find source by following specs: Contact IP: ''; From: '10700'; To: ''; Rline: ''; Contact: ''
11:34:13.299 Endpoint::findSource Can not find source by following specs: Contact IP: ''; From: '10700'; To: ''; Rline: ''; Contact: ''
11:33:58.207 Endpoint::findSource Can not find source by following specs: Contact IP: ''; From: '10700'; To: ''; Rline: ''; Contact: ''
11:33:43.095 Endpoint::findSource Can not find source by following specs: Contact IP: ''; From: '10700'; To: ''; Rline: ''; Contact: ''
11:33:27.973 Endpoint::findSource Can not find source by following specs: Contact IP: ''; From: '10700'; To: ''; Rline: ''; Contact: ''

I don't want to seem ignorant because I have so many concurrent problems. I have scoured the forums and have found many useful tips and guides (thanks itfarmer ) as well as linksys forums with no avail.

Thank you for any advice you have and for the awesome software!

-Tim

 

[Anonymous]

DARN BOY, WHO LET THAT PIG OUT OF THE BARN.

Ok,

If possible send me some screen dumps of your SPA config, and we go from there that gives me a better handle on things.

One thing often forgotten is that even though you have only PSTN you still have to enable LINE 1.

In the Line 1 Tab
1. Dial plan enter something like <#9:><gw0>. #9 being what you dial to get through to the PSTN. You can use another combination if you like.
2. In the PSTN Tab:
Leave dial plan 1 empty
Make sure VoIP-To-PSTN Gateway Enable is set to

 

[N_G]

Also, ensure you follow these steps to make it work with 3CX:

http: //www.3cx.com/voip-gateways/linksys-3102/ - NO LONGER AVAILABLE

 

[ess]

Okay, thanks very much for your help guys.

It is working now!

The problem was that I had an aussie PSTN disconnect tone configured here in the US. I think I followed your config too closely ITFarmer . That was the cause of the frequent disconnects on both ends.

Now, I have a follow-up question. Right now, I use a prefix (9) for dialing numbers on the pstn line, and it works perfectly. However, when testing this system with end users, they complained that they didn't hear a dial tone after pressing 9. Is there any way to make this work?

Furthermore, I have a few different dial plans. Right now, each phone has a dial plan, and the gateway does as well. It works fine with the prefix, I was just wondering if anyone had a suggestion for these areas.

Lastly, I am still getting the following error:

11:02:21.881 Endpoint::findSource Can not find source by following specs: Contact IP: ''; From: '10700'; To: ''; Rline: ''; Contact: ''

This doesn't affect me it doesn't seem, I just wanted to make sure it doesn't mean anything bad.

Other than that, most things are squaring up nicely. A non-vital function I was wondering about was "forwarding to an external number". I have cfwd on the linksys phones as well, however I was wondering how this function works. Does it use conference calling? What I mean, is that if someone calls in on the PSTN line, I understand I won't be able to call right back out on the same gateway pstn line to forward the call to a mobile phone.

Thanks again for all of your help, I appreciate it.

-Tim

 

[jits04]

Create a file, call it: "spa3102.gw.xml" and insert the following rows:

<?xml version="1.0"?>
<doc xmlns:tcx="http://www.3cx.com">
<header>
<type>gateway-template</type>
<version>@PRODUCT_SVN_BUILD@</version>
<time>2007-03-31 18:30:00</time>
</header>

<data>
<device>
<type>gateway</type>
<manufacturer>Sipura</manufacturer>
<model>SPA-3102</model>
<field name="IsInternal">1</field>
<field name="IsBindToMS">0</field>
<field name="UseIPInContact">0</field>
<field name="CallerIDIn">4</field>
<field name="LineType">3</field>
<field name="LineIDIn">3</field>
<field name="CalledNumberIn">2</field>
<field name="DestNumberIn_TO">1</field>
<field name="DestNumberIn_REQLINEURI">1</field>
<field name="DestNumberIn_RPID_CALLED">0</field>
<field name="IsSupportReinvite">1</field>
<field name="IsSupportReplaces">0</field>
<field name="RequireAuthFor">4</field>
<field name="IpInContactReg">1</field>
</device>
</data>
</doc>

Copy it in the \Http\templates. Edit PSTN and apply this template.

Bye

William

 

[Anonymous]

or just download them from here.

http://www.d-d-d.com.au/3cx/Add-Ons/3CXAddOn.htm

Note, these files will soon be moved to another domain (http://www.3cx.com.au)

Just to keep business and pleasure seperated .

 

[nahidul]

In the Line 1 Tab
1. Dial plan enter something like <#9:><gw0>. #9 being what you dial to get through to the PSTN. You can use another combination if you like.
2. In the PSTN Tab:
Leave dial plan 1 empty
Make sure VoIP-To-PSTN Gateway Enable is set to �Yes�
Line 1 VoIP Caller DP: �1�
VoIP Caller Auth Method: �PIN�

Hello.

I have similar problem with my Linksys SPA-3102. Incoming is good, but no outgoing. I can not make any outgoing calls from any of my extensions. I get busy signal after couple seconds. I changed the Line 1 DP and now I am getting the busy signal as soon as I press #8 (I changed DP to use 8 for Linksys SPA-3102, as I have # 9 outbound call rules for VoIP calls).

Please advice.

 

[Anonymous]

I replied to one of your other posts, that might have solved your issue.

if not post again.

We will get to it (eventually) but my local customers and partners get priority sorry.