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New Member

SPA3102 CID not passing to asterisk

Hi,

I have several 3102 for FXO line integration with asterisk, so far so good.

The only thing I was unable to make it work, even with option:

PSTN CID For VoIP CID: yes

Was to make the CID pass to asterisk, if I check the info tab on 3102, in Last PSTN Caller:, I can see the last caller ID number.

What would I be missing?

Thanks in advanced,
Jorge Bastos,

Everyone's tags (1)
1 ACCEPTED SOLUTION

Accepted Solutions
VIP Gold

It seems that "yours" PSTN is

It seems that "yours" PSTN is delivering CID information between first and second ring. Currently, INVITE to Asterisk is sent immediately, e.g. before the CID is known. It needs to be delayed until CID is available.

Move to section "FXO Timer Values".

Set "PSTN Answer Delay" to number (in seconds) that is slightly more than ring period on your PSTN line. Ring period is about 5 second in most countries. so I recommend you use 7 seconds here in such case.

7 REPLIES
VIP Gold

Something is wrong either on

Something is wrong either on SPA3102 side or in Asterisk side. You need to decide. So catch the SIP packets, it will reveal if CID is sent correctly but ignored by Asterisk or not sent by 3102 at all.

New Member

Hi Dan,I'm 100% sure it's

Hi Dan,

I'm 100% sure it's something on 3102 side, it gets the CID, but not passing it to asterisk, on asterisk logs, the CID is blank, here's the output of debug level=3 from 3102:

--
Apr 21 20:18:23 192.168.1.44 FXO:Start CNDD
Apr 21 20:18:24 192.168.1.44 AUD:Stop PSTN Tone
Apr 21 20:18:24 192.168.1.44 AUD:Stop PSTN Tone
Apr 21 20:18:24 192.168.1.44 Calling:pstn@192.168.1.223:0
Apr 21 20:18:24 192.168.1.44 [1:0]AUD ALLOC CALL (port=16390)
Apr 21 20:18:24 192.168.1.44 [1:0]RTP Rx Up
Apr 21 20:18:24 192.168.1.44 CC:Ringback
Apr 21 20:18:24 192.168.1.44 [1:0]RTP Rx Dn
Apr 21 20:18:24 192.168.1.44 AUD:Play PSTN Tone 9
Apr 21 20:18:24 192.168.1.44 CC:Ringback
Apr 21 20:18:24 192.168.1.44 AUD:Play PSTN Tone 9
Apr 21 20:18:24 192.168.1.44 caller id parse number=967874111
Apr 21 20:18:24 192.168.1.44 fxo cnddwrap_feed parse ok 967874111  status=2
Apr 21 20:18:24 192.168.1.44 -- Caller ID: --     Name             = (null) --     Remote Number    = 967874111 --     Dialable Number  = (null) --     No Number Reason = (null) --     No Name Reason   = (null) --     Message Waiting  = (null) --     Date and Time    = 04/21 20:18
Apr 21 20:18:24 192.168.1.44 FXO:CNDD name=, number=967874111
Apr 21 20:18:24 192.168.1.44 FXO:Stop CNDD
Apr 21 20:18:24 192.168.1.44 FXO:CNDD Name= Phone=967874111
Apr 21 20:18:37 192.168.1.44 AUD:Stop PSTN Tone
Apr 21 20:18:37 192.168.1.44 FXO:On Hook
Apr 21 20:18:37 192.168.1.44 AUD:Stop PSTN Tone
Apr 21 20:18:37 192.168.1.44 [0]FM Alert Stop RxTx (c=0025422c;a=0)
Apr 21 20:18:37 192.168.1.44 [1:0]AUD Rel Call
Apr 21 20:18:37 192.168.1.44 DLG Terminated 2e1cf4
Apr 21 20:18:44 192.168.1.44 Sess Terminated
--

And if I play with: PSTN CID For VoIP CID: no

I don't get any CID when calling the FXO line.

Any ideas on which parameter I may be missing?
Been reading them all but don't see anyone that could make sense to change.

VIP Gold

Information I have about the

Information I have about the issue doesn't allow me to make useful advice. Sorry. I wish someone else will help you.

New Member

Here's a full SIP debug from

Here's a full SIP debug from 3102, there's no CID being passed to asterisk.

What info do you need more to help me on this?

 

Apr 22 09:51:57 192.168.1.39
Apr 22 09:52:08 192.168.1.39 FXO:Start CNDD
Apr 22 09:52:09 192.168.1.39 AUD:Stop PSTN Tone
Apr 22 09:52:09 192.168.1.39 AUD:Stop PSTN Tone
Apr 22 09:52:09 192.168.1.39 Calling:234184295@192.168.1.223:0
Apr 22 09:52:09 192.168.1.39 [1:0]AUD ALLOC CALL (port=16428)
Apr 22 09:52:09 192.168.1.39 [1:0]RTP Rx Up
Apr 22 09:52:09 192.168.1.39 [1]->192.168.1.223:5060(991)
Apr 22 09:52:09 192.168.1.39 [1]->192.168.1.223:5060(991)
Apr 22 09:52:09 192.168.1.39 INVITE sip:234184295@192.168.1.223 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.39:5061;branch=z9hG4bK-3db4974^M From: <sip:192.168.1.223>;tag=641077e0787c4294o1^M To: <sip:234184295@192.168.1.223>^M Remote-Party-ID: <sip:192.168.1.223>;screen=yes;party=calling^M Call-ID: 752eaad0-5e822c64@192.168.1.39^M CSeq: 101 INVITE^M Max-Forwards: 70^M Contact: <sip:192.168.1.39:5061>^M Expires: 240^M User-Agent: Linksys/SPA3102-3.3.6(GW)^M Content-Length: 438^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: x-sipura^M Content-Type: application/sdp^M ^M v=0^M o=- 9845 9845 IN IP4 192.168.1.39^M s=-^M c=IN IP4 192.168.1.39^M t=0 0^M m=audio 16428 RTP/AVP 0 2 4 8 18 96 97 98 100 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:2 G726-32/8000^M a=rtpmap:4 G723/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:18 G729a/8000^M a=rtpmap:96 G726-40/8000^M a=rtpmap:97 G726-24/8000^M a=rtpmap:98 G726-16/8000^M a=rtpmap:100 NSE/8000^M a=fmtp:100 192-193^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=ptime:30^M a=sendr
Apr 22 09:52:09 192.168.1.39
Apr 22 09:52:09 192.168.1.39
Apr 22 09:52:09 192.168.1.39 [1]<<192.168.1.223:5060(451)
Apr 22 09:52:09 192.168.1.39 [1]<<192.168.1.223:5060(451)
Apr 22 09:52:09 192.168.1.39 SIP/2.0 100 Trying^M Via: SIP/2.0/UDP 192.168.1.39:5061;branch=z9hG4bK-3db4974;received=192.168.1.39^M From: <sip:192.168.1.223>;tag=641077e0787c4294o1^M To: <sip:234184295@192.168.1.223>^M Call-ID: 752eaad0-5e822c64@192.168.1.39^M CSeq: 101 INVITE^M Server: FPBX-2.11.0(11.8.1)^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M Supported: replaces, timer^M Contact: <sip:234184295@192.168.1.223:5060>^M Content-Length: 0^M ^M
Apr 22 09:52:09 192.168.1.39
Apr 22 09:52:09 192.168.1.39
Apr 22 09:52:09 192.168.1.39 [1]<<192.168.1.223:5060(467)
Apr 22 09:52:09 192.168.1.39 [1]<<192.168.1.223:5060(467)
Apr 22 09:52:09 192.168.1.39 SIP/2.0 180 Ringing^M Via: SIP/2.0/UDP 192.168.1.39:5061;branch=z9hG4bK-3db4974;received=192.168.1.39^M From: <sip:192.168.1.223>;tag=641077e0787c4294o1^M To: <sip:234184295@192.168.1.223>;tag=as1d5df368^M Call-ID: 752eaad0-5e822c64@192.168.1.39^M CSeq: 101 INVITE^M Server: FPBX-2.11.0(11.8.1)^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M Supported: replaces, timer^M Contact: <sip:234184295@192.168.1.223:5060>^M Content-Length: 0^M ^M
Apr 22 09:52:09 192.168.1.39
Apr 22 09:52:09 192.168.1.39
Apr 22 09:52:09 192.168.1.39 CC:Ringback
Apr 22 09:52:09 192.168.1.39 AUD:Play PSTN Tone 9
Apr 22 09:52:09 192.168.1.39 [1:0]RTP Rx Dn
Apr 22 09:52:09 192.168.1.39 [1]<<192.168.1.223:5060(467)
Apr 22 09:52:09 192.168.1.39 [1]<<192.168.1.223:5060(467)
Apr 22 09:52:09 192.168.1.39 SIP/2.0 180 Ringing^M Via: SIP/2.0/UDP 192.168.1.39:5061;branch=z9hG4bK-3db4974;received=192.168.1.39^M From: <sip:192.168.1.223>;tag=641077e0787c4294o1^M To: <sip:234184295@192.168.1.223>;tag=as1d5df368^M Call-ID: 752eaad0-5e822c64@192.168.1.39^M CSeq: 101 INVITE^M Server: FPBX-2.11.0(11.8.1)^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M Supported: replaces, timer^M Contact: <sip:234184295@192.168.1.223:5060>^M Content-Length: 0^M ^M
Apr 22 09:52:09 192.168.1.39
Apr 22 09:52:09 192.168.1.39
Apr 22 09:52:09 192.168.1.39 CC:Ringback
Apr 22 09:52:09 192.168.1.39 AUD:Play PSTN Tone 9
Apr 22 09:52:09 192.168.1.39 FXO:CNDD name=, number=967874111
Apr 22 09:52:09 192.168.1.39 FXO:Stop CNDD
Apr 22 09:52:09 192.168.1.39 FXO:CNDD Name= Phone=967874111
Apr 22 09:52:23 192.168.1.39 AUD:Stop PSTN Tone
Apr 22 09:52:23 192.168.1.39 FXO:On Hook
Apr 22 09:52:23 192.168.1.39 AUD:Stop PSTN Tone
Apr 22 09:52:23 192.168.1.39 FXO:Stop CNDD
Apr 22 09:52:23 192.168.1.39 [0]FM Alert Stop RxTx (c=0023f2a0;a=0)
Apr 22 09:52:23 192.168.1.39 [1:0]AUD Rel Call
Apr 22 09:52:23 192.168.1.39 [1]->192.168.1.223:5060(325)
Apr 22 09:52:23 192.168.1.39 [1]->192.168.1.223:5060(325)
Apr 22 09:52:23 192.168.1.39 CANCEL sip:234184295@192.168.1.223 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.39:5061;branch=z9hG4bK-3db4974^M From: <sip:192.168.1.223>;tag=641077e0787c4294o1^M To: <sip:234184295@192.168.1.223>^M Call-ID: 752eaad0-5e822c64@192.168.1.39^M CSeq: 101 CANCEL^M Max-Forwards: 70^M User-Agent: Linksys/SPA3102-3.3.6(GW)^M Content-Length: 0^M ^M
Apr 22 09:52:23 192.168.1.39
Apr 22 09:52:23 192.168.1.39
Apr 22 09:52:23 192.168.1.39 [1]<<192.168.1.223:5060(433)
Apr 22 09:52:23 192.168.1.39 [1]<<192.168.1.223:5060(433)
Apr 22 09:52:23 192.168.1.39 SIP/2.0 487 Request Terminated^M Via: SIP/2.0/UDP 192.168.1.39:5061;branch=z9hG4bK-3db4974;received=192.168.1.39^M From: <sip:192.168.1.223>;tag=641077e0787c4294o1^M To: <sip:234184295@192.168.1.223>;tag=as1d5df368^M Call-ID: 752eaad0-5e822c64@192.168.1.39^M CSeq: 101 INVITE^M Server: FPBX-2.11.0(11.8.1)^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M Supported: replaces, timer^M Content-Length: 0^M ^M
Apr 22 09:52:23 192.168.1.39
Apr 22 09:52:23 192.168.1.39
Apr 22 09:52:23 192.168.1.39 [1]->192.168.1.223:5060(368)
Apr 22 09:52:23 192.168.1.39 [1]->192.168.1.223:5060(368)
Apr 22 09:52:23 192.168.1.39 ACK sip:234184295@192.168.1.223 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.39:5061;branch=z9hG4bK-3db4974^M From: <sip:192.168.1.223>;tag=641077e0787c4294o1^M To: <sip:234184295@192.168.1.223>;tag=as1d5df368^M Call-ID: 752eaad0-5e822c64@192.168.1.39^M CSeq: 101 ACK^M Max-Forwards: 70^M Contact: <sip:192.168.1.39:5061>^M User-Agent: Linksys/SPA3102-3.3.6(GW)^M Content-Length: 0^M ^M
Apr 22 09:52:23 192.168.1.39
Apr 22 09:52:23 192.168.1.39
Apr 22 09:52:23 192.168.1.39 [1]<<192.168.1.223:5060(417)
Apr 22 09:52:23 192.168.1.39 [1]<<192.168.1.223:5060(417)
Apr 22 09:52:23 192.168.1.39 SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.1.39:5061;branch=z9hG4bK-3db4974;received=192.168.1.39^M From: <sip:192.168.1.223>;tag=641077e0787c4294o1^M To: <sip:234184295@192.168.1.223>;tag=as1d5df368^M Call-ID: 752eaad0-5e822c64@192.168.1.39^M CSeq: 101 CANCEL^M Server: FPBX-2.11.0(11.8.1)^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M Supported: replaces, timer^M Content-Length: 0^M ^M
Apr 22 09:52:23 192.168.1.39
Apr 22 09:52:23 192.168.1.39
Apr 22 09:52:23 192.168.1.39 DLG Terminated 2b62c0

 

VIP Gold

It seems that "yours" PSTN is

It seems that "yours" PSTN is delivering CID information between first and second ring. Currently, INVITE to Asterisk is sent immediately, e.g. before the CID is known. It needs to be delayed until CID is available.

Move to section "FXO Timer Values".

Set "PSTN Answer Delay" to number (in seconds) that is slightly more than ring period on your PSTN line. Ring period is about 5 second in most countries. so I recommend you use 7 seconds here in such case.

New Member

Damn!That's it!I've changed

Damn!

That's it!
I've changed it to 1sec, thanks for your help!

VIP Gold

Glad to hear. Consider

Glad to hear. Consider marking thread as answered. It will help others to found solutions.

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