[triwhdxk]

newbie t o VOIP
I have got SPA3102 and i am using this in UK. My setup is very simple. On Line1 and PSTN i have got 2 accounts from gizmo as all the friends i need to talk to are on gizmo. Both the accounts get registered. Line 1 is on 5060 while Line 2 is 5061

I can call from Line 1 to other gizmo users. I can do #9 to get my PSTN and dial Uk landline from Line 1 phone.

I am attaching my config to this post.

My issue is:

PSTN to VOIP
------------
I call my PSTN and get authenticated through a PIN. Now i hear the VOIP dial tone but it is constant. I dial numbers and nothing happens. Not sure what settings needs to be changed.


VOIP to PSTN
-------------
I call my VOIP and get authenticated through a PIN. When I hear PSTN dial tone and than type the number it dials the PSTN numbers. Now this works OK if a user is calling through a cordless phone or using a gizmo software. If the users uses a normal phone, the user just hears a constant dial tone. Not sure why it works from cordless only.

Appreciate your help in solving this issue.

[hwittenb]

Your normal settings on the PSTN tab look OK for the voip-to-pstn gateway and the pstn-to-voip gateway. On incoming PSTN calls you are doing a ring thru to line 1 with a ringing delay of 10 seconds before the 3102 answers the phone.

You have changed a number of the default telephone interface settings to adapt to your U.K. environment. My guess would be that something there is not correct for your environment.

I would use the SPA's debug server to try to analyze what is happening. Do you see the dtmf digits correctly? Are you getting an immediate disconnect when you start to dial? The debug server output is sometime overwhelming at first but it presents a lot of analytical data. Download the program from the link below, on the Sip Tab set your pc's network ip address, set the debug level to 3, and on the PSTN tab set the debug option to FULL:
SPA adaptors Frequently Asked Questions (FAQ)

[triwhdxk]

For PSTN - VOIP the error i am getting is
Last PSTN Disconnect Reason:CPC Signal

I will use the syslog server to see whats going on.

Any ideas on VOIP to PSTN issue. where it works with cordless and xlite software but not with normal phone

[hwittenb]

Originally Posted by triwhdxk

For PSTN - VOIP the error i am getting is
Last PSTN Disconnect Reason:CPC Signal

Your adapter could be sensing a CPC (Calling Party Control) signal when it takes the FXO port off hook and immediately disconnecting. Try disabling Detect CPC: and see if it makes any difference. If I remember you also had the minimum CPC setting set to 90 ms which is a short time.

Any ideas on VOIP to PSTN issue. where it works with cordless and xlite software but not with normal phone

It is obvious how you use the xlite software. How are you using the cordless and normal phones. How are they making the call over voip?

[triwhdxk]

Here is the syslog for pstn-voip issue
<151>TP Parser error: 34
192.168.0.10 23/08 17:14:21.514
<151>TP Parser error: 34
192.168.0.10 23/08 17:14:21.514
<151>TP Parser error: 34
192.168.0.10 23/08 17:14:49.964
<151>TP Parser error: 34
192.168.0.10 23/08 17:14:49.964
<151>TP Parser error: 34
192.168.0.10 23/08 17:15:18.085
<151>TP Parser error: 34
192.168.0.10 23/08 17:15:18.085
<151>TP Parser error: 34
192.168.0.10 23/08 17:15:47.878
<151>TP Parser error: 34
192.168.0.10 23/08 17:15:47.878
<151>TP Parser error: 34
192.168.0.10 23/08 17:16:15.978
<151>TP Parser error: 34
192.168.0.10 23/08 17:16:15.978
<151>TP Parser error: 34
192.168.0.10 23/08 17:16:46.031
<151>TP Parser error: 34
192.168.0.10 23/08 17:16:46.031
<151>POL REV 50 -51
192.168.0.10 23/08 17:17:05.760
<151>FXO:OnHook PolRev
192.168.0.10 23/08 17:17:05.760
<151>FXO:Start CNDD
192.168.0.10 23/08 17:17:05.760
<151>FXO:CNDD name=UNAVAILABLE, number=
192.168.0.10 23/08 17:17:06.961
<151>FXO:Stop CNDD
192.168.0.10 23/08 17:17:06.961
<159>FXO:CNDD Name=UNAVAILABLE Phone=
192.168.0.10 23/08 17:17:06.971
<151>FXO:Start CNDD
192.168.0.10 23/08 17:17:07.743
<151>Calling:17471961785@127.0.0.1:5060
192.168.0.10 23/08 17:17:08.754
<151>[1:0]AUD ALLOC CALL (port=16396)
192.168.0.10 23/08 17:17:08.754
<151>[1:0]RTP Rx Up
192.168.0.10 23/08 17:17:08.754
<151>[0:0]AUD ALLOC CALL (port=16398)
192.168.0.10 23/08 17:17:08.774
<151>[0:0]RTP Rx Up
192.168.0.10 23/08 17:17:08.774
<151>CC:Ringback
192.168.0.10 23/08 17:17:08.774
<151>AUD:Play PSTN Tone 9
192.168.0.10 23/08 17:17:08.774
<151>[1:0]RTP Rx Dn
192.168.0.10 23/08 17:17:08.784
<151>TP Parser error: 34
192.168.0.10 23/08 17:17:13.861
<151>TP Parser error: 34
192.168.0.10 23/08 17:17:13.861
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:17.747
<159>FXO:Off Hook
192.168.0.10 23/08 17:17:17.747
<151>FXO:Stop CNDD
192.168.0.10 23/08 17:17:17.747
<151>AUD:Play PSTN Tone 19
192.168.0.10 23/08 17:17:17.757
<159>[0]FM Alert Stop RxTx (c=0024fe9c;a=0)
192.168.0.10 23/08 17:17:17.767
<151>[1:0]AUD Rel Call
192.168.0.10 23/08 17:17:17.777
<159>[0]FM Alert Stop RxTx (c=0024a404;a=0)
192.168.0.10 23/08 17:17:17.777
<151>[0:0]AUD Rel Call
192.168.0.10 23/08 17:17:17.777
<151>CC:Ended
192.168.0.10 23/08 17:17:17.787
<151>DLG Terminated 29b1e8
192.168.0.10 23/08 17:17:17.787
<151>DLG Terminated 29b154
192.168.0.10 23/08 17:17:17.867
<151>Sess Terminated
192.168.0.10 23/08 17:17:17.867
<151>Sess Terminated
192.168.0.10 23/08 17:17:17.867
<159>FXOigit=0
192.168.0.10 23/08 17:17:21.653
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:21.753
<159>FXOigit=7
192.168.0.10 23/08 17:17:22.003
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:22.083
<159>FXOigit=7
192.168.0.10 23/08 17:17:22.233
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:22.324
<159>FXOigit=2
192.168.0.10 23/08 17:17:22.484
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:22.574
<159>FXOigit=#
192.168.0.10 23/08 17:17:23.165
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:23.255
<151>AUD:Play PSTN Tone 1
192.168.0.10 23/08 17:17:23.255
<151>AUD:Play PSTN Tone 8
192.168.0.10 23/08 17:17:33.259
<151>FXO:CPC
192.168.0.10 23/08 17:17:39.949
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:39.949
<159>FXO:On Hook
192.168.0.10 23/08 17:17:39.949
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:39.949
<151>FXO:Stop CNDD
192.168.0.10 23/08 17:17:39.949
<151>TP Parser error: 34
192.168.0.10 23/08 17:17:42.703
<151>TP Parser error: 34
192.168.0.10 23/08 17:17:42.713
<151>POL REV 50 -50
192.168.0.10 23/08 17:17:44.245
<151>FXO:OnHook PolRev
192.168.0.10 23/08 17:17:44.245
<151>FXO:Start CNDD
192.168.0.10 23/08 17:17:44.255
<151>FXO:CNDD name=UNAVAILABLE, number=
192.168.0.10 23/08 17:17:45.447
<151>FXO:Stop CNDD
192.168.0.10 23/08 17:17:45.457
<159>FXO:CNDD Name=UNAVAILABLE Phone=
192.168.0.10 23/08 17:17:45.457
<151>FXO:Start CNDD
192.168.0.10 23/08 17:17:46.218
<151>Calling:17471961785@127.0.0.1:5060
192.168.0.10 23/08 17:17:47.229
<151>[1:0]AUD ALLOC CALL (port=16400)
192.168.0.10 23/08 17:17:47.229
<151>[1:0]RTP Rx Up
192.168.0.10 23/08 17:17:47.239
<151>[0:0]AUD ALLOC CALL (port=16402)
192.168.0.10 23/08 17:17:47.249
<151>[0:0]RTP Rx Up
192.168.0.10 23/08 17:17:47.259
<151>CC:Ringback
192.168.0.10 23/08 17:17:47.259
<151>AUD:Play PSTN Tone 9
192.168.0.10 23/08 17:17:47.269
<151>[1:0]RTP Rx Dn
192.168.0.10 23/08 17:17:47.269
<159>[0]Off Hook
192.168.0.10 23/08 17:17:53.218
<151>[1:0]ENC INIT 0
192.168.0.10 23/08 17:17:53.238
<151>[1:0]RTP Tx Up (pt=0->c0a8000a:16402)
192.168.0.10 23/08 17:17:53.248
<151>[1:0]RTCP Tx Up
192.168.0.10 23/08 17:17:53.248
<151>CC:Remote Resume
192.168.0.10 23/08 17:17:53.258
<151>AUD:Play PSTN Tone 9
192.168.0.10 23/08 17:17:53.258
<151>CC:Connected
192.168.0.10 23/08 17:17:53.268
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:53.268
<151>[1:0]RTP Rx Up
192.168.0.10 23/08 17:17:53.268
<151>FXO:PSTN Disconnect Tone
192.168.0.10 23/08 17:17:53.278
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:53.278
<159>FXO:On Hook
192.168.0.10 23/08 17:17:53.288
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:53.288
<151>FXO:Stop CNDD
192.168.0.10 23/08 17:17:53.288
<151>CC:Connected
192.168.0.10 23/08 17:17:53.298
<151>[0:0]ENC INIT 0
192.168.0.10 23/08 17:17:53.298
<151>[0:0]RTP Tx Up (pt=0->c0a8000a:16400)
192.168.0.10 23/08 17:17:53.308
<151>[0:0]RTCP Tx Up
192.168.0.10 23/08 17:17:53.308
<159>[0]FM Alert Stop RxTx (c=0024fe9c;a=0)
192.168.0.10 23/08 17:17:53.318
<151>[1:0]AUD Rel Call
192.168.0.10 23/08 17:17:53.318
<159>FXO:Off Hook
192.168.0.10 23/08 17:17:53.318
<151>FXO:Stop CNDD
192.168.0.10 23/08 17:17:53.328
<151>CC:Ended
192.168.0.10 23/08 17:17:53.328
<159>[0]FM Alert Stop RxTx (c=0024a404;a=0)
192.168.0.10 23/08 17:17:53.338
<151>[0:0]AUD Rel Call
192.168.0.10 23/08 17:17:53.338
<151>DLG Terminated 29b310
192.168.0.10 23/08 17:17:53.338
<151>Sess Terminated
192.168.0.10 23/08 17:17:53.348
<151>DLG Terminated 29b27c
192.168.0.10 23/08 17:17:53.358
<151>Sess Terminated
192.168.0.10 23/08 17:17:53.358
<151>FXO:PSTN Disconnect Tone
192.168.0.10 23/08 17:17:54.169
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:54.169
<159>FXO:On Hook
192.168.0.10 23/08 17:17:54.179
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:54.179
<151>FXO:Stop CNDD
192.168.0.10 23/08 17:17:54.179
<151>FXO:PSTN Disconnect Tone
192.168.0.10 23/08 17:17:54.189
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:54.189
<159>FXO:On Hook
192.168.0.10 23/08 17:17:54.199
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:54.199
<151>FXO:Stop CNDD
192.168.0.10 23/08 17:17:54.199
<151>FXO:PSTN Disconnect Tone
192.168.0.10 23/08 17:17:54.209
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:54.209
<159>FXO:On Hook
192.168.0.10 23/08 17:17:54.219
<151>AUD:Stop PSTN Tone
192.168.0.10 23/08 17:17:54.219
<151>FXO:Stop CNDD
192.168.0.10 23/08 17:17:54.229
<159>[0]Hook Flash
192.168.0.10 23/08 17:17:56.933
<159>[0]Hook Flash
192.168.0.10 23/08 17:17:58.726
<159>[0]On Hook
192.168.0.10 23/08 17:18:01.950

[triwhdxk]

Attach is my current config and syslog.

VOIP TO PSTN:
Using gizmo or xlite it works. Using SIPBROKER access number with cordless phone it works. Using SIPBroker access number with normal phone it does not works and gives PSTN dial tone continuous. Attach is syslog in voip_to_pstn.txt

PSTN TO VOIP:
Calling my PSTN using any phone cordless or normal phone it gives my VOIP dial tone but its continuous. Attach is syslog pstn_to_voip.txt

by the way the current config is with Detect CPC: yes. I have tried Detect CPC: no but still it didn't worked.

appreciate your help.
kashif

[hwittenb]

The TP Parser Error 34 is not clear to me why that happens. I suspect it is some error or bug in the program receiving the data on your pc.

Voip to PSTN
Traces are difficult to understand but I have found that sooner or later the cause of the problem comes to light. It looks like you called several time in the voip to pstn trace. Is this correct?

The voip part answers OK, the problem seems to be receiving the dtmf digits.

It would help me to understand exactly what is happening if I knew exactly the digits you entered and consequently should show on the trace.

Run it again. I would change the authentication to none so that all you are entering is just the number you are trying to dial. Get that working OK and then you can put the authentication back again.

Set the Sip Debug Option to FULL on the PSTN tab


PSTN to Voip

Same comments about understanding traces. I believe the port is being disconnected right at the start after you get the tone. I would try the same thing here, change the authentication to none, increase the data output by setting the Sip Debug Option to FULL on the PSTN tab. Document exactly what you entered on the phone keypad.

[triwhdxk]

I did the full log as you said. removed the PIN both ways. attached is the zip file

PSTN to VOIP
called the PSTN and than tried to dial gizmo test number 17474743246

VOIP to PSTN
called the VOIP number from SIPBroker access number and than on PSTN tried to dial 07944550927 my mobile number. it gave me this message on info tab Last PSTN Disconnect Reason: VoIP Dialing Timeout

[hwittenb]

Originally Posted by triwhdxk

I did the full log as you said. removed the PIN both ways. attached is the zip file

PSTN to VOIP
called the PSTN and than tried to dial gizmo test number 17474743246

The trace doesn't show any digits digitized at the FXO port which is consistent with my theory that the SPA disconnects the FXO port at the start of the call due to CPC detected, or a disconnect tone.

VOIP to PSTN
called the VOIP number from SIPBroker access number and than on PSTN tried to dial 07944550927 my mobile number. it gave me this message on info tab Last PSTN Disconnect Reason: VoIP Dialing Timeout

On the trace I see 045XX0927 where the XX are those TP Parser 34 messages and could be a missing digits. You indicate that you dialed more digits than this which would mean that you are having trouble transporting the dtmf digits thru your voip provider(s). You are using AVT which generally the best way to send the digits.

Removing the pin authentication simplifies the data somewhat.

In both cases, a debug trace with the Debug Level set to 3 on the SipTab, and Sip Debug Option set to FULL on the PSTN tab usually shows a greater level of detail than the trace that you saved. Maybe it has something to do with the speed of your pc in receiving the data over the ethernet link.

[thilina88]

hi my SPA3102 gatway pstn to voip is disconected

Last PSTN Disconnect Reason:VoIP Dialing Timeout