can anyone help me sort out this issue…am not able to view the inbound caller id number routed to freepbx…
I have registered an extension in Xlite soft phone but Caller id displayed as ‘Anonymous’ for inbound calls.
The log in /var/log/asterisk/full should give you everything you need to know about how that happened.
Thanks for the reply Dave, But no luck, i didn’t find any solution in that file 
Look at the CDR for a failed call. (Go to Reports -> CDR Reports, leave default settings, click Search.)
If the caller’s number does not display, the problem is on the trunk side. If you have a VoIP trunk, look at the incoming INVITE to see where the number appears. (At the Asterisk command prompt, do
pjsip set logger on
if pjsip trunk, or
sip set debug on
if using chan_sip.
Then make a failing call and look at the Asterisk log.) If the number doesn’t appear anywhere, you need to open a ticket with the provider.
If you have a POTS (analog) trunk, you need to adjust the caller ID settings for the card or FXO gateway for your country. If you still have trouble, connect an analog phone to the line and confirm that the carrier is sending caller ID.
If the caller’s number does appear in the CDR, the problem is on the extension side. Report what the CDR shows and whether you have the same issue with Zoiper or PhonerLite.
Thanks for the reply Stewart, But we are not using a FXO gateway between analog line and the freePbx, Instead we are using one more pbx i.e. a Epbx called NEC. so FXO gateway is not in the picture.
When calls are routed to NEC Epbx, we are able to view the CallerID number, But when the same calls routed to freepbx through NEC , Caller ID is displayed as ’ ANONYMOUS’.
Even i have tried reading the logs with ‘SIP SET DEBUG ON’ but am unable to figure out where exactly the isuue is…
Do you see a line starting with
INVITE sip:
in the log? If so, there are headers following such as From, Remote-Party-ID or P-Asserted-Identity. If none of them contain the calling number, the problem is on the NEC side. Otherwise, report what you see.
No didn’t find any line starting with ‘INVITE sip:’ , Please have a look at the logs which i have captured…
[2018-05-24 18:18:56] NOTICE[2361]: chan_iax2.c:12039 iax2_poke_peer: Still have a callno…
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
Reliably Transmitting (no NAT) to 10.10.4.215:59698:
OPTIONS sip:[email protected]:59698;rinstance=e80aba4f2b6ea8b5;transport=UDP SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK2fae33b1
Max-Forwards: 70
From: “Unknown” <sip:U [email protected]>;tag=as444bc17f
To: <sip:5011 @X.X.X.X:59698;rinstance=e80aba4f2b6ea8b5;transport=UDP>
Contact: <sip:Unkno [email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.0(1.8.13.0)
Date: Thu, 24 May 2018 12:49:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
Reliably Transmitting (no NAT) toX.X.X.X:57353:
OPTIONS sip:[email protected]:57353;rinstance=4bd0afb94051eafc SIP/2.0
Via: SIP/2.0/UDP 10.10.1.66:5060;branch=z9hG4bK0aeb80fb
Max-Forwards: 70
From: “Unknown” <sip:Unkn [email protected]>;tag=as05ea5a75
To: <sip:[email protected]:573 53;rinstance=4bd0afb94051eafc>
Contact: <sip:Un [email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.0(1.8.13.0)
Date: Thu, 24 May 2018 12:49:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:X.X.X.X:59698 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK2fae33b1
Contact: <sip:X .X.X.X:59698>
To: <sip:[email protected]:59698; rinstance=e80aba4f2b6ea8b5;transport=UDP>;tag=6e0ddc73
From: “Unknown” <sip:Unkno [email protected]>;tag=as444bc17f
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.15.40006 rv2.8.20
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
<— SIP read from UDP:X.X.X.X:57353 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK0aeb80fb
Contact: <sip:X.X .X.X:57353>
To: <sip:[email protected]:5 7353;rinstance=4bd0afb94051eafc>;tag=8fa38852
From: “Unknown” <sip:Unk [email protected]>;tag=as05ea5a75
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Supported: replaces
User-Agent: X-Lite release 5.2.0 stamp 90534
Allow-Events: talk, hold
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to X.X.X.X:5060:
OPTIONS sip:X.X.X.X SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11dd0611;rport
Max-Forwards: 70
From: “Unknown” <sip: Un [email protected]>;tag=as08ba6065
To: <sip:X.X. X.X>
Contact: <sip:Un [email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.0(1.8.13.0)
Date: Thu, 24 May 2018 12:49:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:X.X.X.X:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11dd0611;rport
From: “Unknown” <sip:Unkno [email protected]>;tag=as08ba6065
To: <sip:X.X.X .X>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
Supported: timer
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
[2018-05-24 18:19:16] NOTICE[2360]: chan_iax2.c:12039 iax2_poke_peer: Still have a callno…
<— SIP read from UDP:10.10.4.215:57353 —>
Well, if you have lots of extensions, the call might not be in the last 500 lines (that the GUI displays by default) but you should still find it by searching /var/log/asterisk/full . If it’s not there, make another test call into the system and look again. If it’s still not there, the trunk from the NEC is probably using other than chan_sip. If it’s a pjsip trunk, turn on pjsip logging with
pjsip set logger on
and do
sip set debug off
to cancel chan_sip logging.
Make another test call and check the log file again.
Otherwise, look at the trunk settings and report what technology it’s using (IAX, H.323, etc.)
gave me this reply…
No such command ‘pjsip set logger on’ (type ‘core show help pjsip set’ for other possible commands)
and Finally got the INVITE sip line, here it is:
INVITE sip:50 [email protected]:57353;rinstance=4bd0afb94051eafc SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK4d1e618c
Max-Forwards: 70
From: “Anonymous” <sip:anony [email protected]>;tag=as0b93904b
To: <sip:5013 @X.X.X.X:57353;rinstance=4bd0afb94051eafc>
Contact: <sip: [email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.0(1.8.13.0)
PLEASE help!!!
What you just posted is the INVITE sent from the PBX to the called extension (5013). It indeed shows no caller ID.
Your Asterisk 1.8.x is too old to have pjsip (that’s not a problem, just a comment on why you got the error).
Is the trunk from the NEC using SIP? If so, you need to find the INVITE from the NEC to FreePBX. If none of the headers have the calling number, the problem is at the NEC and you’ll need to fix it there.
If the trunk from NEC is using another technology, please post details.
I think you are expecting these logs:
<— SIP read from UDP: X.X.X.X:5060 —>
INVITE sip:[email protected]. X:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK-2bfa4d239ff92-229
From: “Anonymous” <sip:anonym [email protected]:5060>;tag=7de04d239ff91-229
To: <sip:[email protected] :5060>
Call-ID: VV4_0001.00-26ef98f4-4d2 39ff9 -0-229@ X.X.X.X
CSeq: 1 INVITE
Contact: <sip:a [email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
Max-Forwards: 70
Session-Expires: 1800
Supported: timer
Content-Length: 171
v=0
o=- 27683 0 IN IP4 X.X.X.X
s=-
c=IN IP4 X.X.X.X
t=0 0
m=audio 10166 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=rtpmap:101 telephone-event/8000
Yes, The trunk from NEC is a SIP trunk…
Yes, that is the correct log.
The NEC is not sending the calling number at all. With luck, their documentation is good enough to lead you to an Epbx settings change that will fix the problem. Otherwise, seek help from NEC, the dealer that sold you the system, or a suitable forum. Sorry, I know nothing at all about NEC. Perhaps another member here can help.
Please don’t mention that, Its fine if you don’t know anything about NEC. Thanks a lot for your valuable suggestions Stewart 
And sure i will talk with NEC people and look forward to resolve this issue.
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