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Caller-ID not working anymore

jasch

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jasch Newbie

jasch

Posted
Posted

I remember at some point last year, when I initially bought and configured my CS410, inbound caller-id was working. Not anymore. All calls on the call log appear as from "Anonymous (anonymous@localhost)"

 

I connected a phone directly to the CO line to test it out, and called-id is indeed active and working perfectly, yet my CS410 is not grabbing the info.

 

Any ideas what to look for?

 

Running the latest (3.2.0.3143)

 

Thanks in advance

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Vodia PBX Grand Master

Vodia PBX

Posted
Posted
I remember at some point last year, when I initially bought and configured my CS410, inbound caller-id was working. Not anymore. All calls on the call log appear as from "Anonymous (anonymous@localhost)"

 

I connected a phone directly to the CO line to test it out, and called-id is indeed active and working perfectly, yet my CS410 is not grabbing the info.

 

Any ideas what to look for?

 

Running the latest (3.2.0.3143)

 

What country are you in and what carrier are you using?

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  • 2 weeks later...
Vodia PBX Grand Master

Vodia PBX

Posted
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I am in Costa Rica. I am using the only carrier available (state monopoly). I can connect my Panasonic phone (bought on Amazon) directly to the phone jack, and I get caller id info.

 

You should change the PSTN gateway logging flag (admin->logging settings), set the log level fairly high (e.g. 9) and reboot the device. Then make an inbound call and check the log for information on what the gateway sees regarding caller-ID. Maybe Costa Rica is the next country that had it's own idea on how a caller-ID should be presented...

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jasch Newbie

jasch

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You should change the PSTN gateway logging flag (admin->logging settings), set the log level fairly high (e.g. 9) and reboot the device. Then make an inbound call and check the log for information on what the gateway sees regarding caller-ID. Maybe Costa Rica is the next country that had it's own idea on how a caller-ID should be presented...

 

I just did. I have nothing on the logs regarding PSTN activity. In fact, I turned everything OFF on the logs, except PSTN GATEWAY logging. Nothing on the logs (even tough I called from my mobile phone to the PSTN, and got connected to a friends extension on the US).

 

So, the PSTN, and the CS410 is working ok. Just no PSTN logging and no caller-id.

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jasch Newbie

jasch

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Here's the call coming in through the PSTN ports (127.0.0.1)

 

And the call going out to the extension:

 

[7] 2009/04/17 16:01:15:	SIP Rx udp:127.0.0.1:5062:
INVITE sip:22830598@localhost;user=phone SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062
From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=756898537
To: <sip:22830598@localhost;user=phone>
Call-ID: e4b2d6b4@fxo
Contact: <sip:127.0.0.1:5062;line=1>
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 137

v=0
o=root 0 0 IN IP4 1.1.1.2
s=-
c=IN IP4 1.1.1.2
t=0 0
m=audio 2060 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=ptime:20

 

Besides this, there's nothing else regarding any PSTN gateway info

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  • 3 weeks later...
jasch Newbie

jasch

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So no answers at all in 3 weeks. Gotta tell you, the box is pretty great. A little pricier, but hey I wanted something solid and trouble free. Trouble free it hasn't been, and support is extremely poor. The wiki is lacking in so many areas, and support from pbxnsip is almost non-existant, besides a couple of replies that never amount to real answers.

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pbx support Newbie

pbx support

Posted
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So no answers at all in 3 weeks. Gotta tell you, the box is pretty great. A little pricier, but hey I wanted something solid and trouble free. Trouble free it hasn't been, and support is extremely poor. The wiki is lacking in so many areas, and support from pbxnsip is almost non-existant, besides a couple of replies that never amount to real answers.

 

Have you set the log level to 9 and taken the log? You should see PSTN logs when you receive/make PSTN calls

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jasch Newbie

jasch

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I have set the log level to 9 and taken the log? You should see PSTN logs when you receive/make PSTN calls

 

As I explained 3 messages above, I did, and there's no PSTN information in the logs at all. Even tough PSTN is being used. Just the irrelevant info I pasted above.

 

This is how I have it configured:

post-609-1241471420_thumb.png

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pbx support Newbie

pbx support

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As I explained 3 messages above, I did, and there's no PSTN information in the logs at all. Even tough PSTN is being used. Just the irrelevant info I pasted above.

 

This is how I have it configured:

 

Can you set the "Log Other messages (e.g. INVITE)" to "yes" and take the log. Thanks

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jasch Newbie

jasch

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Can you set the "Log Other messages (e.g. INVITE)" to "yes" and take the log. Thanks

 

Attached.

 

[7] 2009/05/05 11:23:25:	SIP Tr udp:127.0.0.1:5062:
SIP/2.0 183 Ringing
Via: SIP/2.0/UDP 127.0.0.1:5062
From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=407487131
To: <sip:22830598@localhost;user=phone>;tag=0598e82c74
Call-ID: 925643fd@fxo
CSeq: 1 INVITE
Contact: <sip:22830598@201.198.25.110:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.2.0.3143
Content-Type: application/sdp
Content-Length: 216

v=0
o=- 269702622 269702622 IN IP4 201.198.25.110
s=-
c=IN IP4 201.198.25.110
t=0 0
m=audio 16456 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[9] 2009/05/05 11:23:28:	Resolve 248834: udp 69.244.126.113 41306
[9] 2009/05/05 11:23:29:	Resolve 248835: udp 69.244.126.113 33945
[8] 2009/05/05 11:23:32:	DNS: dns_naptr callcentric.com expired
[8] 2009/05/05 11:23:32:	DNS: dns_srv _sips._tcp.callcentric.com expired
[8] 2009/05/05 11:23:32:	DNS: dns_srv _sip._tcp.callcentric.com expired
[7] 2009/05/05 11:23:33:	SIP Tr udp:127.0.0.1:5062:
SIP/2.0 183 Ringing
Via: SIP/2.0/UDP 127.0.0.1:5062
From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=407487131
To: <sip:22830598@localhost;user=phone>;tag=0598e82c74
Call-ID: 925643fd@fxo
CSeq: 1 INVITE
Contact: <sip:22830598@201.198.25.110:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.2.0.3143
Content-Type: application/sdp
Content-Length: 216

v=0
o=- 269702622 269702622 IN IP4 201.198.25.110
s=-
c=IN IP4 201.198.25.110
t=0 0
m=audio 16456 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[9] 2009/05/05 11:23:34:	Remote site closed the connection
[9] 2009/05/05 11:23:37:	Last message repeated 4 times
[9] 2009/05/05 11:23:37:	Resolve 248836: url sip:callcentric.com
[9] 2009/05/05 11:23:37:	Resolve 248836: naptr callcentric.com
[8] 2009/05/05 11:23:37:	DNS: Add dns_naptr callcentric.com (ttl=60)
[9] 2009/05/05 11:23:37:	Resolve 248836: naptr callcentric.com
[9] 2009/05/05 11:23:37:	Resolve 248836: srv tls _sips._tcp.callcentric.com
[8] 2009/05/05 11:23:37:	Attendant: Timeout (extension)
[8] 2009/05/05 11:23:37:	DNS: Add dns_srv _sips._tcp.callcentric.com (ttl=60)
[9] 2009/05/05 11:23:37:	Resolve 248836: srv tls _sips._tcp.callcentric.com
[9] 2009/05/05 11:23:37:	Resolve 248836: srv tcp _sip._tcp.callcentric.com
[9] 2009/05/05 11:23:37:	Resolve 248837: udp 69.244.126.113 33945
[7] 2009/05/05 11:23:37:	SIP Tx udp:69.244.126.113:33945:
CANCEL sip:609@69.244.126.113:33945 SIP/2.0
Via: SIP/2.0/UDP 201.198.25.110:5060;branch=z9hG4bK-ea4120266e1eb4a20434cd783dd9c55f;rport
From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1575693843
To: "Nicolas Musmanni" <sip:609@localhost>
Call-ID: f8cdbd4e@pbx
CSeq: 22699 CANCEL
Max-Forwards: 70
Content-Length: 0

[5] 2009/05/05 11:23:37:	Attendant: Redirecting to external number 19089105034
[8] 2009/05/05 11:23:37:	Play audio_en/pb_hold_redirect.wav
[9] 2009/05/05 11:23:37:	Resolve 248838: aaaa udp 127.0.0.1 5062
[9] 2009/05/05 11:23:37:	Resolve 248838: a udp 127.0.0.1 5062
[9] 2009/05/05 11:23:37:	Resolve 248838: udp 127.0.0.1 5062
[7] 2009/05/05 11:23:37:	SIP Tx udp:127.0.0.1:5062:
SIP/2.0 200 Ringing
Via: SIP/2.0/UDP 127.0.0.1:5062
From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=407487131
To: <sip:22830598@localhost;user=phone>;tag=0598e82c74
Call-ID: 925643fd@fxo
CSeq: 1 INVITE
Contact: <sip:22830598@201.198.25.110:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.2.0.3143
Content-Type: application/sdp
Content-Length: 216

v=0
o=- 269702622 269702622 IN IP4 201.198.25.110
s=-
c=IN IP4 201.198.25.110
t=0 0
m=audio 16456 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[7] 2009/05/05 11:23:37:	SIP Rx udp:127.0.0.1:5062:
ACK sip:22830598@localhost;user=phone SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062
From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=407487131
To: <sip:22830598@localhost;user=phone>
Call-ID: 925643fd@fxo
Contact: <sip:127.0.0.1:5062;line=1>
CSeq: 1 ACK
Content-Length: 0

[8] 2009/05/05 11:23:37:	DNS: Add dns_srv _sip._tcp.callcentric.com (ttl=60)
[9] 2009/05/05 11:23:37:	Resolve 248836: srv tcp _sip._tcp.callcentric.com
[9] 2009/05/05 11:23:37:	Resolve 248836: srv udp _sip._udp.callcentric.com
[9] 2009/05/05 11:23:37:	Resolve 248836: a udp alpha1.callcentric.com 5080
[9] 2009/05/05 11:23:37:	Resolve 248836: udp 204.11.192.22 5080
[9] 2009/05/05 11:23:37:	Resolve 248836: a udp alpha7.callcentric.com 5080
[9] 2009/05/05 11:23:37:	Resolve 248836: udp 204.11.192.37 5080
[9] 2009/05/05 11:23:37:	Resolve 248836: a udp alpha2.callcentric.com 5080
[9] 2009/05/05 11:23:37:	Resolve 248836: udp 204.11.192.23 5080
[9] 2009/05/05 11:23:37:	Resolve 248836: a udp callcentric.com 5060
[9] 2009/05/05 11:23:37:	Resolve 248836: udp 204.11.192.22 5060
[8] 2009/05/05 11:23:37:	Trunk 2 (callcentric) has outbound proxy udp:204.11.192.22:5060 udp:204.11.192.22:5080 udp:204.11.192.23:5080 udp:204.11.192.37:5080
[9] 2009/05/05 11:23:37:	Resolve 248839: url sip:callcentric.com
[9] 2009/05/05 11:23:37:	Resolve 248839: naptr callcentric.com
[9] 2009/05/05 11:23:37:	Resolve 248839: srv tls _sips._tcp.callcentric.com
[9] 2009/05/05 11:23:37:	Resolve 248839: srv tcp _sip._tcp.callcentric.com
[9] 2009/05/05 11:23:37:	Resolve 248839: srv udp _sip._udp.callcentric.com
[9] 2009/05/05 11:23:37:	Resolve 248839: a udp alpha7.callcentric.com 5080
[9] 2009/05/05 11:23:37:	Resolve 248839: udp 204.11.192.37 5080
[7] 2009/05/05 11:23:37:	Receiving DTMF on codec 101
[7] 2009/05/05 11:23:37:	SIP Rx udp:69.244.126.113:33945:
SIP/2.0 487 Request Terminated
To: "Nicolas Musmanni" <sip:609@localhost>;tag=e86eb70450804f44i0
From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1575693843
Call-ID: f8cdbd4e@pbx
CSeq: 22699 INVITE
Via: SIP/2.0/UDP 201.198.25.110:5060;branch=z9hG4bK-ea4120266e1eb4a20434cd783dd9c55f
Server: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0

[7] 2009/05/05 11:23:37:	Call f8cdbd4e@pbx#1575693843: Clear last INVITE
[9] 2009/05/05 11:23:37:	Resolve 248840: url sip:69.244.126.113:33945;transport=udp
[9] 2009/05/05 11:23:37:	Resolve 248840: a udp 69.244.126.113 33945
[9] 2009/05/05 11:23:37:	Resolve 248840: udp 69.244.126.113 33945
[7] 2009/05/05 11:23:37:	SIP Tx udp:69.244.126.113:33945:
ACK sip:609@69.244.126.113:33945 SIP/2.0
Via: SIP/2.0/UDP 201.198.25.110:5060;branch=z9hG4bK-ea4120266e1eb4a20434cd783dd9c55f;rport
From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1575693843
To: "Nicolas Musmanni" <sip:609@localhost>;tag=e86eb70450804f44i0
Call-ID: f8cdbd4e@pbx
CSeq: 22699 ACK
Max-Forwards: 70
Contact: <sip:609@201.198.25.110:5060;transport=udp>
Content-Length: 0

[5] 2009/05/05 11:23:37:	INVITE Response 487 Request Terminated: Terminate f8cdbd4e@pbx
[7] 2009/05/05 11:23:37:	Other Ports: 1
[7] 2009/05/05 11:23:37:	Call Port: 925643fd@fxo#0598e82c74
[7] 2009/05/05 11:23:37:	SIP Rx udp:69.244.126.113:33945:
SIP/2.0 200 OK
To: "Nicolas Musmanni" <sip:609@localhost>;tag=e86eb70450804f44i0
From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1575693843
Call-ID: f8cdbd4e@pbx
CSeq: 22699 CANCEL
Via: SIP/2.0/UDP 201.198.25.110:5060;branch=z9hG4bK-ea4120266e1eb4a20434cd783dd9c55f
Server: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0

[8] 2009/05/05 11:23:37:	Answer challenge with username 17772288450
[9] 2009/05/05 11:23:37:	Resolve 248841: udp 204.11.192.37 5080 udp:1
[9] 2009/05/05 11:23:37:	Message repetition, packet dropped
[9] 2009/05/05 11:23:37:	Resolve 248842: aaaa udp 204.11.192.37 5080
[9] 2009/05/05 11:23:37:	Resolve 248842: a udp 204.11.192.37 5080
[9] 2009/05/05 11:23:37:	Resolve 248842: udp 204.11.192.37 5080
[5] 2009/05/05 11:23:40:	Attendant: Redirecting to 19089105034
[9] 2009/05/05 11:23:40:	Dialplan: Evaluating !^1([0-9]*)@.*!sip:1\1@\r;user=phone!i against 19089105034@localhost
[5] 2009/05/05 11:23:40:	Dialplan 609: Match 19089105034@localhost to <sip:19089105034@sip.voipbuster.com;user=phone> on trunk voipbuster
[5] 2009/05/05 11:23:40:	Using "Anonymous" <sip:anonymous@localhost;user=phone> as redirect source address
[8] 2009/05/05 11:23:40:	Play audio_moh/noise.wav
[9] 2009/05/05 11:23:40:	UDP: Opening socket on port 16464
[9] 2009/05/05 11:23:40:	UDP: Opening socket on port 16465
[9] 2009/05/05 11:23:40:	Resolve 248843: url sip:19089105034@sip.voipbuster.com;user=phone
[9] 2009/05/05 11:23:40:	Resolve 248843: naptr sip.voipbuster.com
[8] 2009/05/05 11:23:40:	DNS: Add dns_naptr sip.voipbuster.com (ttl=60)
[9] 2009/05/05 11:23:40:	Resolve 248843: naptr sip.voipbuster.com
[9] 2009/05/05 11:23:40:	Resolve 248843: srv tls _sips._tcp.sip.voipbuster.com
[8] 2009/05/05 11:23:40:	DNS: Add dns_srv _sips._tcp.sip.voipbuster.com (ttl=60)
[9] 2009/05/05 11:23:40:	Resolve 248843: srv tls _sips._tcp.sip.voipbuster.com
[9] 2009/05/05 11:23:40:	Resolve 248843: srv tcp _sip._tcp.sip.voipbuster.com
[8] 2009/05/05 11:23:40:	DNS: Add dns_srv _sip._tcp.sip.voipbuster.com (ttl=60)
[9] 2009/05/05 11:23:40:	Resolve 248843: srv tcp _sip._tcp.sip.voipbuster.com
[9] 2009/05/05 11:23:40:	Resolve 248843: srv udp _sip._udp.sip.voipbuster.com
[8] 2009/05/05 11:23:41:	DNS: Add dns_srv _sip._udp.sip.voipbuster.com (ttl=60)
[9] 2009/05/05 11:23:41:	Resolve 248843: srv udp _sip._udp.sip.voipbuster.com
[9] 2009/05/05 11:23:41:	Resolve 248843: a udp sip.voipbuster.com 5060
[9] 2009/05/05 11:23:41:	Resolve 248843: udp 194.120.0.198 5060
[7] 2009/05/05 11:23:41:	SIP Tx udp:194.120.0.198:5060:
INVITE sip:19089105034@sip.voipbuster.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 201.198.25.110:5060;branch=z9hG4bK-c89e47443d707d2d2c2a834fb96ff912;rport
From: <sip:jasch98@sip.voipbuster.com>;tag=307949700
To: <sip:19089105034@sip.voipbuster.com;user=phone>
Call-ID: 7f70ab16@pbx
CSeq: 28247 INVITE
Max-Forwards: 70
Contact: <sip:jasch98@201.198.25.110:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.2.0.3143
Content-Type: application/sdp
Content-Length: 278

v=0
o=- 494684503 494684503 IN IP4 201.198.25.110
s=-
c=IN IP4 201.198.25.110
t=0 0
m=audio 16464 RTP/AVP 0 8 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[7] 2009/05/05 11:23:41:	SIP Rx udp:194.120.0.198:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 201.198.25.110:5060;branch=z9hG4bK-c89e47443d707d2d2c2a834fb96ff912;rport
From: <sip:jasch98@sip.voipbuster.com>;tag=307949700
To: <sip:19089105034@sip.voipbuster.com;user=phone>
Contact: sip:19089105034@194.120.0.198:5060
Call-ID: 7f70ab16@pbx
CSeq: 28247 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip.voipbuster.com",nonce="1220037171",algorithm=MD5
Content-Length: 0

[8] 2009/05/05 11:23:41:	Answer challenge with username jasch98
[9] 2009/05/05 11:23:41:	Resolve 248844: udp 194.120.0.198 5060 udp:1
[7] 2009/05/05 11:23:41:	SIP Tx udp:194.120.0.198:5060:
ACK sip:19089105034@sip.voipbuster.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 201.198.25.110:5060;branch=z9hG4bK-c89e47443d707d2d2c2a834fb96ff912;rport
From: <sip:jasch98@sip.voipbuster.com>;tag=307949700
To: <sip:19089105034@sip.voipbuster.com;user=phone>
Call-ID: 7f70ab16@pbx
CSeq: 28247 ACK
Max-Forwards: 70
Content-Length: 0

[9] 2009/05/05 11:23:41:	Resolve 248845: udp 194.120.0.198 5060 udp:1
[7] 2009/05/05 11:23:41:	SIP Tx udp:194.120.0.198:5060:
INVITE sip:19089105034@sip.voipbuster.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 201.198.25.110:5060;branch=z9hG4bK-dce6c7633471b868395b99ae218ef8a7;rport
From: <sip:jasch98@sip.voipbuster.com>;tag=307949700
To: <sip:19089105034@sip.voipbuster.com;user=phone>
Call-ID: 7f70ab16@pbx
CSeq: 28248 INVITE
Max-Forwards: 70
Contact: <sip:jasch98@201.198.25.110:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.2.0.3143
Authorization: Digest realm="sip.voipbuster.com",nonce="1220037171",response="332cc19e0db586346b7a7a72ad27006a",username="jasch98",uri="sip:19089105034@sip.voipbuster.com;user=phone",algorithm=MD5
Content-Type: application/sdp
Content-Length: 278

v=0
o=- 494684503 494684503 IN IP4 201.198.25.110
s=-
c=IN IP4 201.198.25.110
t=0 0
m=audio 16464 RTP/AVP 0 8 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[9] 2009/05/05 11:23:41:	Message repetition, packet dropped
[7] 2009/05/05 11:23:41:	SIP Rx udp:194.120.0.198:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 201.198.25.110:5060;branch=z9hG4bK-dce6c7633471b868395b99ae218ef8a7;rport
From: <sip:jasch98@sip.voipbuster.com>;tag=307949700
To: <sip:19089105034@sip.voipbuster.com;user=phone>
Contact: sip:19089105034@194.120.0.198:5060
Call-ID: 7f70ab16@pbx
CSeq: 28248 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

[7] 2009/05/05 11:23:41:	SIP Rx udp:194.120.0.198:5060:
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 201.198.25.110:5060;branch=z9hG4bK-dce6c7633471b868395b99ae218ef8a7;rport
From: <sip:jasch98@sip.voipbuster.com>;tag=307949700
To: <sip:19089105034@sip.voipbuster.com;user=phone>;tag=c41710acc42b10ac49edd93f642282
Contact: sip:19089105034@194.120.0.198:5060
Call-ID: 7f70ab16@pbx
CSeq: 28248 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 198

v=0
o=jasch98 1241544220 1241544220 IN IP4 62.41.83.179
s=SIP Call
c=IN IP4 62.41.83.179
t=0 0
m=audio 41736 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
[7] 2009/05/05 11:23:41:	Set packet length to 20
[6] 2009/05/05 11:23:41:	Sending RTP for 7f70ab16@pbx#307949700 to 62.41.83.179:41736
[7] 2009/05/05 11:23:41:	7f70ab16@pbx#307949700: RTP pass-through mode
[7] 2009/05/05 11:23:41:	925643fd@fxo#0598e82c74: RTP pass-through mode
[7] 2009/05/05 11:23:41:	Receiving DTMF on codec 101

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Vodia PBX Grand Master

Vodia PBX

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There should be something in the log with "PSTN:" in it. Check the logging page if you have turned PSTN event logging on (you may turn other events off to reduce the number of messages that you see in the log file) and make sure that the log level is high enough; after that you need to reboot the device before it takes effect.

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jasch Newbie

jasch

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There should be something in the log with "PSTN:" in it. Check the logging page if you have turned PSTN event logging on (you may turn other events off to reduce the number of messages that you see in the log file) and make sure that the log level is high enough; after that you need to reboot the device before it takes effect.

 

My logging settings are just as in the image I posted two messages above. Except per your recommendation, now it has Log Other messages (e.g. INVITE) set to "yes"

 

I'll reboot the unit and give it another try. Does it has to be a hard reboot (turn power switch on/off) or a restart of the unit is fine (form the System/IP-Setub RESTART button)?

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jasch Newbie

jasch

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My logging settings are just as in the image I posted two messages above. Except per your recommendation, now it has Log Other messages (e.g. INVITE) set to "yes"

 

I'll reboot the unit and give it another try. Does it has to be a hard reboot (turn power switch on/off) or a restart of the unit is fine (form the System/IP-Setub RESTART button)?

 

I did a hard reboot. With the settings on the image included.

 

I called the phone number attached to the CS410, the phone hooked up to the telephone adapter rang for about 10 seconds. I hang up.

 

When I go check the log, the log is empty.

 

It's what I've been trying to tell you guys for about 3 weeks. Even tough I check PSTN logging, nothing is logged from the PSTN (even tough It's working).

post-609-1241546502_thumb.png

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Vodia PBX Grand Master

Vodia PBX

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I did a hard reboot. With the settings on the image included.

 

I called the phone number attached to the CS410, the phone hooked up to the telephone adapter rang for about 10 seconds. I hang up.

 

When I go check the log, the log is empty.

 

It's what I've been trying to tell you guys for about 3 weeks. Even tough I check PSTN logging, nothing is logged from the PSTN (even tough It's working).

 

Hmm. This is strange. Can you log into the box (SSH) and check what is in /etc/sipfxo.conf? There should be a line with "SIPLOG", and it should be set to "true". If not, just use "vi" to edit that file and give it another reboot...

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jasch Newbie

jasch

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Hmm. This is strange. Can you log into the box (SSH) and check what is in /etc/sipfxo.conf? There should be a line with "SIPLOG", and it should be set to "true". If not, just use "vi" to edit that file and give it another reboot...

 

Sure. What is the SSH password?

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jasch Newbie

jasch

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Thw default is root123

 

You were right. That settings was set to "false" so no matter how many times I modified my log settings, there would never be any PSTN log entries.

 

I changed the settings, rebooted, and here's the log.

[3] 2009/05/07 20:03:42:	PSTN: Channel 0 going to RING
[9] 2009/05/07 20:03:47:	UDP: Opening socket on port 16462
[9] 2009/05/07 20:03:47:	UDP: Opening socket on port 16463
[5] 2009/05/07 20:03:47:	PSTN: Response code: 100
[9] 2009/05/07 20:03:47:	UDP: Opening socket on port 16448
[9] 2009/05/07 20:03:47:	UDP: Opening socket on port 16449
[5] 2009/05/07 20:03:47:	PSTN: Response code: 183
[3] 2009/05/07 20:03:47:	PSTN: Channel 0 going to NO_RING
[5] 2009/05/07 20:03:48:	PSTN: Response code: 183
[5] 2009/05/07 20:03:49:	Last message repeated 2 times
[3] 2009/05/07 20:03:49:	PSTN: Channel 0 going to RING
[5] 2009/05/07 20:03:49:	PSTN: Ringing, but last invite = 1
[5] 2009/05/07 20:03:51:	PSTN: Response code: 183
[3] 2009/05/07 20:03:53:	PSTN: Channel 0 going to NO_RING
[5] 2009/05/07 20:03:55:	PSTN: Response code: 183
[3] 2009/05/07 20:03:56:	PSTN: Channel 0 going to RING
[5] 2009/05/07 20:03:56:	PSTN: Ringing, but last invite = 1
[5] 2009/05/07 20:03:57:	PSTN: Response code: 200
[5] 2009/05/07 20:03:57:	PSTN: RTP destination=6e19c6c9
[5] 2009/05/07 20:03:57:	PSTN: RTP destination=16462
[5] 2009/05/07 20:03:57:	PSTN: RTP OOB codec=101
[6] 2009/05/07 20:03:57:	PSTN: Start call on 0
[5] 2009/05/07 20:03:57:	PSTN: Channel 0 goes offhook
[3] 2009/05/07 20:03:57:	PSTN: Channel 0 going to TALKING
[5] 2009/05/07 20:03:57:	PSTN: Country Code set to 64
[5] 2009/05/07 20:03:57:	PSTN: Tone Detection set to 0
[5] 2009/05/07 20:04:19:	PSTN: Received BYE message on channel 0
[3] 2009/05/07 20:04:19:	PSTN: Channel 0: Hangup
[5] 2009/05/07 20:04:19:	PSTN: Channel 0 goes onhook
[5] 2009/05/07 20:04:19:	PSTN: enable_callerid 0
[3] 2009/05/07 20:04:19:	PSTN: Channel 0 going to GO_ONHOOK
[3] 2009/05/07 20:04:20:	PSTN: Channel 0 going to IDLE

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There was some issue in saving the PSTN related configuration on CS410. Probably, that's why you could not save it. This version should take care of the saving issue http://pbxnsip.com/cs410/update-3.3.2.3181.tgz

 

 

 

You were right. That settings was set to "false" so no matter how many times I modified my log settings, there would never be any PSTN log entries.

 

I changed the settings, rebooted, and here's the log.

[3] 2009/05/07 20:03:42:	PSTN: Channel 0 going to RING
[9] 2009/05/07 20:03:47:	UDP: Opening socket on port 16462
[9] 2009/05/07 20:03:47:	UDP: Opening socket on port 16463
[5] 2009/05/07 20:03:47:	PSTN: Response code: 100
[9] 2009/05/07 20:03:47:	UDP: Opening socket on port 16448
[9] 2009/05/07 20:03:47:	UDP: Opening socket on port 16449
[5] 2009/05/07 20:03:47:	PSTN: Response code: 183
[3] 2009/05/07 20:03:47:	PSTN: Channel 0 going to NO_RING
[5] 2009/05/07 20:03:48:	PSTN: Response code: 183
[5] 2009/05/07 20:03:49:	Last message repeated 2 times
[3] 2009/05/07 20:03:49:	PSTN: Channel 0 going to RING
[5] 2009/05/07 20:03:49:	PSTN: Ringing, but last invite = 1
[5] 2009/05/07 20:03:51:	PSTN: Response code: 183
[3] 2009/05/07 20:03:53:	PSTN: Channel 0 going to NO_RING
[5] 2009/05/07 20:03:55:	PSTN: Response code: 183
[3] 2009/05/07 20:03:56:	PSTN: Channel 0 going to RING
[5] 2009/05/07 20:03:56:	PSTN: Ringing, but last invite = 1
[5] 2009/05/07 20:03:57:	PSTN: Response code: 200
[5] 2009/05/07 20:03:57:	PSTN: RTP destination=6e19c6c9
[5] 2009/05/07 20:03:57:	PSTN: RTP destination=16462
[5] 2009/05/07 20:03:57:	PSTN: RTP OOB codec=101
[6] 2009/05/07 20:03:57:	PSTN: Start call on 0
[5] 2009/05/07 20:03:57:	PSTN: Channel 0 goes offhook
[3] 2009/05/07 20:03:57:	PSTN: Channel 0 going to TALKING
[5] 2009/05/07 20:03:57:	PSTN: Country Code set to 64
[5] 2009/05/07 20:03:57:	PSTN: Tone Detection set to 0
[5] 2009/05/07 20:04:19:	PSTN: Received BYE message on channel 0
[3] 2009/05/07 20:04:19:	PSTN: Channel 0: Hangup
[5] 2009/05/07 20:04:19:	PSTN: Channel 0 goes onhook
[5] 2009/05/07 20:04:19:	PSTN: enable_callerid 0
[3] 2009/05/07 20:04:19:	PSTN: Channel 0 going to GO_ONHOOK
[3] 2009/05/07 20:04:20:	PSTN: Channel 0 going to IDLE

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Vodia PBX Grand Master

Vodia PBX

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I updated the firmware, yet still not caller id information. Now that I have PSTN log entries, what could be wrong?

 

Well, the log tells us, that the PBX is able to detect that the line is ringing; but no caller-ID; not in FSK mode nor in DTMF mode....... AFAIK there are no other modes on his planet.

 

So you are sure that e.g. a phone is able to show the Caller-ID? If that is a yea, the only idea I have left is that you might need to amplify the signal; maybe it is not strong enough and we have an analog problem (do you have a good volume on the line?).

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jasch Newbie

jasch

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So you are sure that e.g. a phone is able to show the Caller-ID? If that is a yea, the only idea I have left is that you might need to amplify the signal; maybe it is not strong enough and we have an analog problem (do you have a good volume on the line?).

 

After your message, I had my doubts. Called the phone company, everything is in order. Check the volume on the line, it's great (it comes directly from the phone box), I can even make data calls without dropping packets using that line.

 

I tried 3 different phones, including a cheap as hell from the local costco $5 phone. They all worked. How come the CS410 is the only one that doesn't? Here's a picture of the Panasonic DECT 6.0 phone I got from Amazon.com last month, showing CallerID informacion.

 

The only thing I haven't tried is reversing the polarity of the incoming CO line, but I don't see how that could affect.

post-609-1242079355_thumb.jpg

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Vodia PBX Grand Master

Vodia PBX

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So? Any explanations as to why caller id works on every device I plug in except the cs410? Obvioulsy there's something the cs410 is NOT doing properly...

 

:P I wish we had wireshark for FXO. The only thing we can do is try to nail the problem down with a couple of gateway builds that give us more logging information (will probably take a few loops).

 

If you can, please load a special build from http://pbxnsip.com/protect/sipfxo-cid-1, and load it into the /pbx directory of the PBX (e.g. using psftp.exe), move the old sipfxo away (e.g. mv sipfxo sipfxo.old.1), then rename sipfxo-cid-1 to sipfxo and restart the box. There should be more logging available that tells us where it sets the caller-ID to Anonymous.

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