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Unable to dial via gw0

Hi all,
I have not touched my SPA3102 in a while.
It sits there doing it's thing and once a year I change it to forward calls from PSTN to VOIP to my email.

Sadly at my age, my memory is not that great either. :D
However I have found out I cannot dial out by GW0 for calls I do not get charged for.

 

My dial plan is

(x | 100S0<:@gw0> | 999S0<:@gw0> | 112S0<:@gw0> | 15[01]S0<:@gw0> | 1[45]71<:@gw0> |    0[58]0xx.<:@gw0> | 00x. | <0:0044>[12]x. |         <0:0044>[67]x. | 084x. | <087:004487>x.| 030x.|  033x. | 034x. | <:00441792> [2-9]xxxxx |  <#0,:>xx.<:@draytel.org> | <#1,:>[x*]             [x*].<:@sipbroker.com> | <#2,:>[x*][x*].<:@gw2> | <#3,:>[x*][x*].<:@gw3> | <#4,:>[x*]  [x*].<:@gw4> | <#90,:141>xx.<:@gw0>|<#91:08081701701S0><:@gw0> |   <#92:08081708708S0><:@gw0> | 118! | 09!)

A syslog shows.

syslog server(port:514) started on Thu Jul 26 19:19:20 2018
[1]->217.14.138.127:5060(400)
[1]->217.14.138.127:5060(400)
NOTIFY sip:draytel.org SIP/2.0

Via: SIP/2.0/UDP 82.7.48.193:15061;branch=z9hG4bK-2fa48192;rport

From: Paul Steel <sip:8210859@draytel.org>;tag=297c0aab3fc731a9o1

To: <sip:draytel.org>

Call-ID: c2c706c3-a8aea471@192.168.2.99

CSeq: 21569 NOTIFY

Max-Forwards: 70

Contact: Paul Steel <sip:8210859@82.7.48.193:15061>

Event: keep-alive

User-Agent: Linksys/SPA3102-5.1.10(GW)

Content-Length: 0





[1]<<217.14.138.127:5060(350)
[1]<<217.14.138.127:5060(350)
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.2.99:15061;branch=z9hG4bK-2fa48192;rport=15061

From: Paul Steel <sip:8210859@draytel.org>;tag=297c0aab3fc731a9o1

To: <sip:draytel.org>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.756d

Call-ID: c2c706c3-a8aea471@192.168.2.99

CSeq: 21569 NOTIFY

Server: OpenSIPS (1.5.3-notls (x86_64/linux))

Content-Length: 0





[0]Off Hook
2. Report digit 1 (1)(40 ms)
2. Report digit 5 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
Calling:150@127.0.0.1:15061
[0:0]AUD ALLOC CALL (port=16442)
[0:0]RTP Rx Up
[0]->127.0.0.1:15061(884)
[0]->127.0.0.1:15061(884)
INVITE sip:150@127.0.0.1:15061 SIP/2.0

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-9f5b3a62;rport

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

To: <sip:150@127.0.0.1:15061>

Remote-Party-ID: Paul Steel <sip:8210859@draytel.org>;screen=yes;privacy=full;party=calling

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 101 INVITE

Max-Forwards: 70

Contact: Anonymous <sip:anonymous@192.168.2.99:15060>

Expires: 240

User-Agent: Linksys/SPA3102-5.1.10(GW)

Content-Length: 257

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

Supported: x-sipura, replaces

Content-Type: application/sdp



v=0

o=- 32500258 32500258 IN IP4 192.168.2.99

s=-

c=IN IP4 192.168.2.99

t=0 0

m=audio 16442 RTP/AVP 0 100 101

a=rtpmap:0 PCMU/8000

a=rtpmap:100 NSE/8000

a=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv



[1]<<127.0.0.1:15060(884)
[1]<<127.0.0.1:15060(884)
INVITE sip:150@127.0.0.1:15061 SIP/2.0

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-9f5b3a62;rport

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

To: <sip:150@127.0.0.1:15061>

Remote-Party-ID: Paul Steel <sip:8210859@draytel.org>;screen=yes;privacy=full;party=calling

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 101 INVITE

Max-Forwards: 70

Contact: Anonymous <sip:anonymous@192.168.2.99:15060>

Expires: 240

User-Agent: Linksys/SPA3102-5.1.10(GW)

Content-Length: 257

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

Supported: x-sipura, replaces

Content-Type: application/sdp



v=0

o=- 32500258 32500258 IN IP4 192.168.2.99

s=-

c=IN IP4 192.168.2.99

t=0 0

m=audio 16442 RTP/AVP 0 100 101

a=rtpmap:0 PCMU/8000

a=rtpmap:100 NSE/8000

a=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv



[1]->127.0.0.1:15060(321)
[1]->127.0.0.1:15060(321)
SIP/2.0 100 Trying

To: <sip:150@127.0.0.1:15061>

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 101 INVITE

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-9f5b3a62;received=127.0.0.1;rport=15060

Server: Linksys/SPA3102-5.1.10(GW)

Content-Length: 0





[1:0]AUD ALLOC CALL (port=16444)
[1:0]RTP Rx Up
[1]->127.0.0.1:15060(474)
[1]->127.0.0.1:15060(474)
SIP/2.0 180 Ringing

To: <sip:150@127.0.0.1:15061>;tag=4df355a94dd91b0ei1

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 101 INVITE

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-9f5b3a62;received=127.0.0.1;rport=15060

Contact: Paul Steel <sip:150@192.168.2.99:15061>

Server: Linksys/SPA3102-5.1.10(GW)

Remote-Party-ID: Paul Steel <sip:8210859@draytel.org>;screen=yes;party=called

Content-Length: 0





AUD:Stop PSTN Tone
[0]<<127.0.0.1:15061(321)
[0]<<127.0.0.1:15061(321)
SIP/2.0 100 Trying

To: <sip:150@127.0.0.1:15061>

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 101 INVITE

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-9f5b3a62;received=127.0.0.1;rport=15060

Server: Linksys/SPA3102-5.1.10(GW)

Content-Length: 0





[0]<<127.0.0.1:15061(474)
[0]<<127.0.0.1:15061(474)
SIP/2.0 180 Ringing

To: <sip:150@127.0.0.1:15061>;tag=4df355a94dd91b0ei1

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 101 INVITE

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-9f5b3a62;received=127.0.0.1;rport=15060

Contact: Paul Steel <sip:150@192.168.2.99:15061>

Server: Linksys/SPA3102-5.1.10(GW)

Remote-Party-ID: Paul Steel <sip:8210859@draytel.org>;screen=yes;party=called

Content-Length: 0





CC:Ringback
[0:0]RTP Rx Dn
[1]->127.0.0.1:15060(853)
[1]->127.0.0.1:15060(853)
SIP/2.0 200 OK

To: <sip:150@127.0.0.1:15061>;tag=4df355a94dd91b0ei1

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 101 INVITE

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-9f5b3a62;received=127.0.0.1;rport=15060

Contact: Paul Steel <sip:150@192.168.2.99:15061>

Server: Linksys/SPA3102-5.1.10(GW)

Remote-Party-ID: Paul Steel <sip:8210859@draytel.org>;screen=yes;party=called

Content-Length: 257

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

Supported: x-sipura, replaces

Content-Type: application/sdp



v=0

o=- 32500260 32500260 IN IP4 192.168.2.99

s=-

c=IN IP4 192.168.2.99

t=0 0

m=audio 16444 RTP/AVP 0 100 101

a=rtpmap:0 PCMU/8000

a=rtpmap:100 NSE/8000

a=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv



[0]<<127.0.0.1:15061(853)
[0]<<127.0.0.1:15061(853)
SIP/2.0 200 OK

To: <sip:150@127.0.0.1:15061>;tag=4df355a94dd91b0ei1

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 101 INVITE

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-9f5b3a62;received=127.0.0.1;rport=15060

Contact: Paul Steel <sip:150@192.168.2.99:15061>

Server: Linksys/SPA3102-5.1.10(GW)

Remote-Party-ID: Paul Steel <sip:8210859@draytel.org>;screen=yes;party=called

Content-Length: 257

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

Supported: x-sipura, replaces

Content-Type: application/sdp



v=0

o=- 32500260 32500260 IN IP4 192.168.2.99

s=-

c=IN IP4 192.168.2.99

t=0 0

m=audio 16444 RTP/AVP 0 100 101

a=rtpmap:0 PCMU/8000

a=rtpmap:100 NSE/8000

a=fmtp:100 192-193

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv



[0]->192.168.2.99:15061(413)
[0]->192.168.2.99:15061(413)
ACK sip:150@192.168.2.99:15061 SIP/2.0

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-5d8f8208;rport

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

To: <sip:150@127.0.0.1:15061>;tag=4df355a94dd91b0ei1

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 101 ACK

Max-Forwards: 70

Contact: Anonymous <sip:anonymous@192.168.2.99:15060>

User-Agent: Linksys/SPA3102-5.1.10(GW)

Content-Length: 0





[0:0]ENC INIT 0
[0:0]RTP Tx Up (pt=0->c0a80263:16444)
[0:0]RTCP Tx Up
CC:Remote Resume
CC:Connected
[0:0]RTP Rx Up
[1]<<127.0.0.1:15060(413)
[1]<<127.0.0.1:15060(413)
ACK sip:150@192.168.2.99:15061 SIP/2.0

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-5d8f8208;rport

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

To: <sip:150@127.0.0.1:15061>;tag=4df355a94dd91b0ei1

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 101 ACK

Max-Forwards: 70

Contact: Anonymous <sip:anonymous@192.168.2.99:15060>

User-Agent: Linksys/SPA3102-5.1.10(GW)

Content-Length: 0





CC:Connected
AUD:Stop PSTN Tone
[1:0]ENC INIT 0
[1:0]RTP Tx Up (pt=0->c0a80263:16442)
[1:0]RTCP Tx Up
FXO:Off Hook
FXO:Stop CNDD
[0:0]RTP Rx 1st PKT @16442(3)
[1:0]RTP Rx 1st PKT @16444(3)
[1:0]DEC INIT 0
[0:0]DEC INIT 0
[1:0]RTP Dst Change:7f000001:16442
[0:0]RTP Dst Change:7f000001:16444
[1]->217.14.138.127:5060(400)
[1]->217.14.138.127:5060(400)
NOTIFY sip:draytel.org SIP/2.0

Via: SIP/2.0/UDP 82.7.48.193:15061;branch=z9hG4bK-e293aadf;rport

From: Paul Steel <sip:8210859@draytel.org>;tag=297c0aab3fc731a9o1

To: <sip:draytel.org>

Call-ID: c2c706c3-a8aea471@192.168.2.99

CSeq: 21570 NOTIFY

Max-Forwards: 70

Contact: Paul Steel <sip:8210859@82.7.48.193:15061>

Event: keep-alive

User-Agent: Linksys/SPA3102-5.1.10(GW)

Content-Length: 0





[1]<<217.14.138.127:5060(350)
[1]<<217.14.138.127:5060(350)
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.2.99:15061;branch=z9hG4bK-e293aadf;rport=15061

From: Paul Steel <sip:8210859@draytel.org>;tag=297c0aab3fc731a9o1

To: <sip:draytel.org>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.430e

Call-ID: c2c706c3-a8aea471@192.168.2.99

CSeq: 21570 NOTIFY

Server: OpenSIPS (1.5.3-notls (x86_64/linux))

Content-Length: 0





[0]On Hook
[0]FM Alert Stop RxTx (c=0024e5e8;a=0)
[0:0]AUD Rel Call
[0]->192.168.2.99:15061(358)
[0]->192.168.2.99:15061(358)
BYE sip:150@192.168.2.99:15061 SIP/2.0

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-815edae2;rport

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

To: <sip:150@127.0.0.1:15061>;tag=4df355a94dd91b0ei1

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 102 BYE

Max-Forwards: 70

User-Agent: Linksys/SPA3102-5.1.10(GW)

Content-Length: 0





[1]<<127.0.0.1:15060(358)
[1]<<127.0.0.1:15060(358)
BYE sip:150@192.168.2.99:15061 SIP/2.0

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-815edae2;rport

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

To: <sip:150@127.0.0.1:15061>;tag=4df355a94dd91b0ei1

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 102 BYE

Max-Forwards: 70

User-Agent: Linksys/SPA3102-5.1.10(GW)

Content-Length: 0





[1]->127.0.0.1:15060(337)
[1]->127.0.0.1:15060(337)
SIP/2.0 200 OK

To: <sip:150@127.0.0.1:15061>;tag=4df355a94dd91b0ei1

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 102 BYE

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-815edae2;received=127.0.0.1;rport=15060

Server: Linksys/SPA3102-5.1.10(GW)

Content-Length: 0





CC:Ended
AUD:Stop PSTN Tone
[0]FM Alert Stop RxTx (c=002550b0;a=0)
[1:0]AUD Rel Call
AUD:Stop PSTN Tone
FXO:On Hook
AUD:Stop PSTN Tone
FXO:Stop CNDD
AUD:Stop PSTN Tone
[0]<<127.0.0.1:15061(337)
[0]<<127.0.0.1:15061(337)
SIP/2.0 200 OK

To: <sip:150@127.0.0.1:15061>;tag=4df355a94dd91b0ei1

From: Anonymous <sip:anonymous@localhost>;tag=3c564cc351e7febao0

Call-ID: 998f3837-44ce9d6@localhost

CSeq: 102 BYE

Via: SIP/2.0/UDP 192.168.2.99:15060;branch=z9hG4bK-815edae2;received=127.0.0.1;rport=15060

Server: Linksys/SPA3102-5.1.10(GW)

Content-Length: 0





DLG Terminated 2ea2c8
Sess Terminated
DLG Terminated 2ea1a0
Sess Terminated

I am please to see a Howard Wittenburg here, who I believe has helped me immensely in another forum.

My username was Gasman on that forum.

 

The result on the info screen is as below after the call.


pstn line.PNG

7 REPLIES 7
Highlighted
Participant

Paul,

Are you confident you have no problem with the PSTN line? Can you attach a phone directly to the cable attaching to the FXO port and make calls to your numbers?

 

The SPA3102 INFO screen you posted says the SPA3102 is still connected to the PSTN line, still off hook, but your text says it was after the call. The PSTN line voltage should be about 48v when the line is idle, not 7v and off hook as shown on the screen.

 

I compared your trace to a trace of a similiar call on my SPA3102 and my trace looked pretty similiar to yours. The sip trace shows you are trying to call 150 on the attached PSTN line. I believe the trace shows that it connected to the FXO port successfully but apparently you heard no sound. Neither trace actually showed the dial digits going to the off hook PSTN line but apparently that is not something the trace shows. The traces seem to indicate that sound is being transferred by rtp and the info screen indicates rtp bytes were sent and received.

 

A key setting on the PSTN Line Tab is One Stage Dialing: Yes,  however I think you have that set correctly or you would get a dial tone.

 

Highlighted

Hi Howard,

Hope you are well, and thank you for the reply.

AFAIK nothing has changed. The SPA3102 goes off on the weekend via a timer as I get free calls on the weekend. I will try the phone setup tomorrow.

 

I *think* it is something I changed, but I cannot think what. The only time I generally change something is when I go on holiday and have it send me any incoming calls as attachments. I tried to go to the voxilla forum to see a post I seem to recall making and fixing myself, to which you replied but that forum appears to have been taken down. Certainly the PSTN line is fine as I have been using it this weekend without any problems.

 

I must admit I did not think of physical line problems, just the adapter.

 

Will report back.

Highlighted

Howard,

The PSTN line works fine when the unit is switched off (on weekends).

Tonight I unplugged the lead from the phone, the other end of which goes in to the phone port of the adapter. I then unplugged the line lead from the adapter and plugged that directly into the phone and dialled my mobile. No problems there.? I believe that rules out the lead and the PSTN line.?

 

Would it help if I saved my configuration and posted that.? 

I've attached it below. It seems this site doe snot allow the actual html, so have printed as a pdf.

Highlighted

Attempt to upload pdf, but think it will be too large.

Highlighted

Paul,

Thanks for checking out the phone line. From the trace it looked like was should be working OK but I couldn't think of an additional problem.

 

A copy of your configuration could help give me new ideas. If you can't attach the configuration that you saved with your browser then I don't know how to send it on this forum. I can't deal with your .pdf copy. The .pdf copy was a printed copy of the configuration's .html code.

 

As an alternative, email the old fashioned copy that you saved with your browser to your hard drive to my temporary email address: hwharbor-cisco@yahoo.com

 

A couple of quesions:

When you make the failed call do you hear anything at all?

What is the voltage level of the PSTN line when it is idle, on-hook?

Howard

Highlighted

Trace discloses no issue - line go off-hook, then dialing, then connected, then on-hook. Everything looks OK.

 

But records have no time stamps, so issues related to timing may not be apparent from it.

Highlighted

Hi Dan,

 

Thanks for the reply.

It was actually my stupidity and bad memory being the cause.

Howard Wittenburg pretty much spotted it straightaway when he had access to the config.

His response was

 

'The PSTN line dial plan would be the Line 1 VoIP Caller DP: which says 8 under VoIP-To-PSTN Gateway Setup.  Dial Plan 8 is (S0<:8210859@draytel.org>) which I believe what you use to send a call to your voicemail.  Anyway 8210859@draytel.org won't work when you want to dial on the PSTN line.  I think the Line 1 VoIP Caller DP should be 1.'

 

That was the cause. The config gets changed for when I go on holiday, so as to get any calls as voicemail. I had forgotten to set it back to 1. As I don't make that many calls, it has taken since March to come to light.

Whilst I doubt anyone else would get this problem, I thought I'd post it for information knowledge, if nothing else.