Hi,
I am able to capture the Caller ID in SIP Debug but I have no knowledge how to extract it.
Therefore the pbx shows Anonymous from all incoming calls.
The SIP provider require the PBX to use P-Asserted-Identity to display the Caller Line Identity (CLIP).
I have tried the context provided by freepbx such as [from-pstn-toheader] but it did not extract the CLIP from the Sip Header.
Please advise.
Equipment Version
FPBX-13.0.190.7(13.12.1)
Asterisk 13.12.1
Testing scenario
+60399999999 calling +60388888888
Outcome
DID (+60388888888) is detected correctly via PAI and sent to Extension
But incoming Calling Line Identity appears as Anonymous.
The following is the output of the SIP debug
<--- SIP read from UDP:XX.XX.XX.XX:5060 --->
INVITE tel:+60388888888 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK0byz189xyt118w8sxzy121x90;Role=3;Dpt=7862_36;TRC=ffffffff-ffffffff
Route: <sip:sATXX.XX.XX.XX:5160;lr;ann=VOBB_1120>
Record-Route: <sip:XX.XX.XX.XX;lr;Dpt=7862_36;Role=3;CxtId=4;spln=P;X-HwB2bUaCookie=15734;TRC=ffffffff-ffffffff>
Call-ID: xj8plg1gipp0zlr2i8a2vijli88j2a38ATATS.rcatshw01.ims.sipprovider.com.151
From: <sip:AnonymousATims.xxxxx>;tag=r1gijhja-CC-151
To: <sip:88888888ATims.xxxxx>
From: <sip:AnonymousATims.xxxxxxxx>;tag=r1gijhja-CC-151
To: <sip:88888888ATims.xxxxxxxx>
CSeq: 1 INVITE
Alert-Info: info=pattern1
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER
Contact: <sip:**+60399999999**ATXX.XX.XX.XX:5060>
Max-Forwards: 62
Supported: 100rel,timer
Session-Expires: 1200
Min-SE: 1200
P-Called-Party-ID: <tel:+60388888888>
P-Access-Network-Info: xDSL;dsl-location="XXX_V1032 eth 0/11/0/13:400";"location-info=XXX_V1032 eth 0/11/0/13:400"
P-Early-Media: gated
Content-Length: 219
Content-Type: application/sdp
v=0
o=HuaweiATS9900 5313141 5313141 IN IP4 XX.XX.XX.XX
s=Sip Call
c=IN IP4 XX.XX.XX.XX
t=0 0
m=audio 41170 RTP/AVP 8 0 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
<------------->
--- (20 headers 11 lines) ---
Sending to XX.XX.XX.XX:5060 (NAT)
Sending to XX.XX.XX.XX:5060 (NAT)
Using INVITE request as basis request - xj8plg1gipp0zlr2i8a2vijli88j2a38 ATATS.rcatshw01.ims.sipprovider.com.151
Found peer 'multisip' for 'Anonymous' from XX.XX.XX.XX:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port XX.XX.XX.XX:41170
Looking for +60388888888 in from-pstn (domain )
sip_route_dump: route/path hop: <sip:XX.XX.XX.XX;lr;Dpt=7862_36;Role=3;CxtId=4;spln=P;X-HwB2bUaCookie=15734;TRC=ffffffff-ffffffff>