Cannot connect to PSTN SPA3102

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retailx

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I cannot receive calls or make calls using my PSTN line that is connect to my SPA3102 (LAN192.168.176.2) My 3CX server is on LAN:192.168.176.146 ;

I can make internal extension calls. The 3CX line status for all extensions (including the SPA extention) ae showing as registered (green) and idle. The PSTN line (10000) 6868808 is showing as registered (idel) and green.

This is my PSTN line configuration of 3CX:

External number: 6868808
Internal number: 10000
Outbound Caller ID: 6868808
Allow outbound calls: Yes
Allow incomming calls: Yes
Answer After: 0
Authentication ID: 10000
Pass: 10000
Route inbound call to Ext.100

On the SPA3102, the PSTN Line is showing as registered.

When I call the PSTN line from my mobile it just rings out. The information on the SPA3102 shows that the call is being received. It shows up my mobile number as being the last number to call so I know that the SPA3102 is getting the call. However, it does nothing with the call.

I have set-up dial plan as S0:10000 and made sure I have referenced this dial-plan. The following is taken from SPA3102 information page:

PSTN Line Status
Hook State: On Line Voltage: 56 (V)
Loop Current: 0.0 (mA) Registration State: Registered
Last Registration At: 1/1/2003 12:28:06 Next Registration In: 1 s
Last Called VoIP Number: Last Called PSTN Number:
Last VoIP Caller: Last PSTN Caller: , 0862853886
Last PSTN Disconnect Reason: PSTN Activity Timer: 30000 (ms)
Mapped SIP Port: 5060 Call Type:
Yahoo!:
VoIP State: Idle PSTN State: Idle


When I made an outgoing call from extension 100, I dial 96868880 and I get a timeout error. I have included the server logs below for the various actions. Note that when I try to call the PSTN that the 3CX server does not log anything. That is, the SPA3102 does not appear to be trying to connect to the 3CX server at all.

SPA3102 Sip Settings
Sip Transprt UDP
Sip Port 5060
SIP 100REL Enable Yes
EXT SIP Port 5060
Auto Resync Reboot Yes
SIP Proxy Require <Blank>
SIP Remote Party ID Yes
SIP GUID Yes

I have read and configured all settings as per useful guides posted here but am at my wits end!
 
Server Log

Sorry I should have included the server log that is created when I trial to dial out using prefix 9 to connect to the registered PSTN. It reads:

15:59:08.438 CallConf::Rejected Call (C:2A) is rejected: Destination is not answering
15:59:08.438 StratInOut::onCancel Call from Ext.100 to 6868880 has been terminated; reason: Request Timeout
15:58:36.218 CallConf::onIncoming Incoming call from Ext.100 to sip:[email protected]

There is no log created in the 3CX server for calls into the PSTN line.
 
http://www.dayboro.info/3cx/Take2/NetworkView.htm

Have you followed the above?

There are the screen dumps of my SPA3102 and it seems that it helps a lot of people to get it registred and connected..

Henk.
 
Re: Server Log

retailx said:
Sorry I should have included the server log that is created when I trial to dial out using prefix 9 to connect to the registered PSTN. It reads:

15:59:08.438 CallConf::Rejected Call (C:2A) is rejected: Destination is not answering
15:59:08.438 StratInOut::onCancel Call from Ext.100 to 6868880 has been terminated; reason: Request Timeout
15:58:36.218 CallConf::onIncoming Incoming call from Ext.100 to sip:[email protected]

There is no log created in the 3CX server for calls into the PSTN line.


Like wewake mentioned the PSTN and Line1 port numbers should be different. but I just noticed in your log that the destination is not answering?

Is there a dial tone when you call, is the SPA actually making the call? Dialtones are different per countries and needs some special attention every now and than.

what country are getting out of bed in the morning to make a call from ?
 
Getting out of bed in Ireland. I followed the settings as per excellent online guide. Perhaps there needs to be small changes for Ireland. I expect UK settings should be the same.

Thanks for the help. I will check port settings for Line 1 when I get back to office later today. However, I think line 1 is not enabled as I am not trying to connect to a VOIP server as yet. Just trying to connect to the PSTN connected to the Linksys

Just one other issue. Does anyone know what the correct blinking LED lights should be on the Linksys gateway should be? I think that the ON/POWER LED light is not constantly on and is blinking - is this normal?
 
I double checked Henks online manual and made some changes based on these screenshots. I not see the registered PSTN line becoming active when I dial 9 to access it. However, still cannot make outgoing calls on the PSTN. I am getting this from the 3CX server log:

14:41:17.946 StratLink::onHangUp Call(C:36): got Hang-Up from Ln:10000@PSTN; reason: BYE
14:40:45.912 CallLegImpl::onConnected Established media channel for Ln:10000@PSTN: remote=127.0.0.1:16446; local=192.168.176.182:10414
14:40:45.896 StratInOut::onConnected Call from Ext.100 to Ln:10000@PSTN is established

Is remote=127.0.0.1 correct?

I apologise for this and hope I am not wasting other users time!
 
Is remote=127.0.0.1 correct?
That will be the "standard" ip address of the box. Do a ping or tracert to that ip number and you will find that it is your box.


I apologise for this and hope I am not wasting other users time!
Do not worry about that, we gonna make an expert out of you :-).

Ok back to the problem at hand.

Looks like you can make a call to the PSTN but for no clear reason the call gets dropped after 30 odd seconds. Time is important so if you can confirm that this happens everytime that would be good.

Now what can this be... the line hangs up. Few reasons why that can happen.

1. the caller or called party hang up, this is not happening in your case.
2. the SPA things you hang up and detected a silent, this initiates a disconnect.


Check your settings that they are correct for Ireland, especially
PSTN Disconnect Detection section (the silent might be set to 30 try to set this to 10 and make a call than set it to 60 and make a call post both logs).

International Control

The above settings are unique per country and they will control some of the call progress tones.

With the silent detection I like to esblish if a call is actually made or if there is just silent on the line and therefore a call gets dropped. Make sure you talk into the handset whilst doing the tests. (well atleast everbody else things yo got it working :-)) I just want to make sure the sound comes through.


Perhaps there needs to be small changes for Ireland.
The 3CX guys are going even one better, there is a part of the forum dedicated to settings especially VOIP providers. I assume that will be a good section to collect the different settings.

The more the better so we can build a knowledge base on what does and does not work.

Hope this helps
henk


Henk.
 
With the following SPA3102 settings under PSTN Disconnect Detection

Detect CPC: yes
Detect Polarity Reversal: yes
Detect PSTN Long Silence: no
Detect VoIP Long Silence: no
PSTN Long Silence Duration: 10
VoIP Long Silence Duration: 30
PSTN Silence Threshold: medium
Min CPC Duration: 0.2
Detect Disconnect Tone: Yes
Disconnect Tone: 480@-30,620@-30;4(.25/.25/1+2)

This is the 3CX server log:
10:25:02.745 StratLink::onHangUp Call(C:B): got Hang-Up from Ln:10000@PSTN; reason: BYE
10:24:30.769 CallLegImpl::onConnected Established media channel for Ln:10000@PSTN: remote=127.0.0.1:16386; local=192.168.176.182:10648
10:24:30.769 StratInOut::onConnected Call from Ext.100 to Ln:10000@PSTN is established
10:24:30.769 CallLegImpl::onConnected Established media channel for Ext.100: remote=192.168.176.182:10648; local=127.0.0.1:16386
10:24:30.644 CallConf::onIncoming Incoming call from Ext.100 to sip:[email protected]

Changing the PSTN Long Silence Duration to 60 made no difference with the following 3CX server output:
10:27:20.851 StratLink::onHangUp Call(C:C): got Hang-Up from Ln:10000@PSTN; reason: BYE
10:27:10.947 StunClient::process STUN resolved external IP=86.43.108.58:55530 by server 194.221.62.209
10:26:48.890 CallLegImpl::onConnected Established media channel for Ln:10000@PSTN: remote=127.0.0.1:16388; local=192.168.176.182:10862
10:26:48.875 StratInOut::onConnected Call from Ext.100 to Ln:10000@PSTN is established
10:26:48.875 CallLegImpl::onConnected Established media channel for Ext.100: remote=192.168.176.182:10862; local=127.0.0.1:16388
10:26:48.765 CallConf::onIncoming Incoming call from Ext.100 to sip:[email protected]

Note that after a select OK on the Sip hardphone I get what sounds like a dialtone for .25 of a second and then there is silence.

In relation to the IP address 127.0.0.1 I pinged it and am getting a reply from that address: However, my LAN ip address of the SPA3102 is set to 192.168.176. - I thought that 127.0.0.1 is what is known as a loopback connection. That is if one tries to connect to 127.0.0.1 then one is actually trying to connect to your own machine.

Again, thanks for your help
 
Re 127.0.0.1 your correct, you can change that in the hosts table.

Interesting to see that your call gets disconnected after 32 sec no matter what.

In the provisioning area change these settings. (currently they are 3600)
Resync Random Delay: Set to 500
Resync Periodic: set to 500

See how you go with that, i am trying to enforce a quick re-register.

If that does not work we have to set each regional setting unique to Ireland, that will take me a while.:-)

Henk.
 
Henk

thanks for fast reply.

I made those changes and the connection still gets dropped after 32 seconds.

In terms of the 127.0.0.1 issue - is this an issue? When you say I can change in hosts tables what do you mean?
 
No no leave the 127.0.0.1 as it is, one step at the time mate.

I have to dive into the books on this one :-). But it is 8:30 pm here in aus so Ill pack up and spend some time with the fam.

I will lookinto this on the weekend, unless someone else has a better idea. Not sure why this call is dropping at this stage. For sure it will be something simple i overlook at the moment.

Sorry, I have to get back to you on this one.

Henk.
 
Henk

do not know if this helps but even after a hang-up on the extention trying to get out through the PSTN the connection on 3CX server continues up to the 32 seconds before the ext->PSTN gateway is disconnected.

It would appear as if the hang-up on the voip phone is being ignored....
 
32 seconds before the ext->PSTN gateway is disconnected
That can be the disconnect tone in the pstn line.

The setting you used is for australia :-). (move up here and the bloody thing will work for sure LOL).

480@-30,620@-30;4(.25/.25/1+2) now I do not dream these things up. This one is the standard out of the box one.

Also introduce the PSTN Silence Threshold, this works as follows.
Choose from {very low, low, medium, high, very high}. The lower the setting, the easier to detect silence and hence easier to trigger a disconnection. (one would expect it was the otherway arround actually)

Detect PSTN Long Silence set to yes for both VoIP and PSTN.

This is minimum length of PSTN silence (or inactivity) in seconds to trigger a gateway call disconnection if <Detect Long Silence> is
 
Henk

thanks for that.

things have got worse for me!!! Playing with settings I have managed to lose the link that I had between IP phone/3CX/SPA - when I trial to dial out from phone I now get the following:

11:31:11.851 CallConf::Rejected Call (C:7) is rejected: User Not Found
11:31:11.851 StratInOut::onCancel Call from Ext.100 to 6868880 has been terminated; reason: User Not Found
11:31:11.741 Endpoint::findSource Can not find source by following specs: From: '10000'; To: '6868880'; Rline: '6868880'; Contact: '10000'
11:31:11.662 CallConf::onIncoming Incoming call from Ext.100 to sip:[email protected]

I have my dial plan set-up properly. The PSTN and the line I am calling from are showing as registered. Now I am not getting through to the Linksys at all for some reason.

At this stage I am getting on the next plane to Sydney!
 
Ok I HAD ENOUGH OF THIS. :lol:

You gonna be my "special pet" project.

Can you do me a favour and send me the screen dumpsof you SPa setting. (use something like SnagIT to get the screens you can use the trial version i think).
And all the other info you have, I do not need account details etc but if you can tell me how it all is connected that will help.


Reason why I am asking is that I need to have a "browse" through the screens and see how it all hangs together.

I will PM you my email address so you can send it to that.

It is something silly for sure, but I feel I we are getting of track so lets have a closer look :-).

Henk.

PS. if you come this way, get of the plane one stop before sydney otherwise you gonna be up for a 24 hrs drive to my place (LOL).
 
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