Failed to Make Calls from TE/TB to SIP trunk When Caller ID is Blank

Symptom

In some cases, the ISDN PBX/Line would send blank caller ID to our gateways. If you could login the SSH and Asterisk CLI, you could find the logs like the following:

You would the there is no caller ID behind the "from".

Refer this guide to enter the Asterisk CLI and get the logs: Asterisk CLI

-- Accepting overlap call from '' to '0412345678' on channel 0/12, span 2

-- Starting simple switch on 'DAHDI/12-1'

Although the call flow is successful to dial out by SIP trunk, but the the SIP Trunk provider returns 403, 404 response or other fatal response to gateways.

If you enable the SIP  debug, you would see the following SIP packets. That gateway is trying to send the anonymous caller ID.

(* In the log, we hidden the public IP and use fake number)

Reliably Transmitting (NAT) to *.*.*.*:5060:
INVITE sip:0412345678@ipvoice.one:5060 SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK5836c158;rport
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as1407029f
To: <sip:0412345678@ipvoice.one:5060>
Contact: <sip:Anonymous@*.*.*.*>
Call-ID: 0c46d23d4f8e683f4a10d1f74d327899@ipvoice.one
CSeq: 102 INVITE
User-Agent: TE100
Date: Mon, 11 Feb 2019 02:35:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 814698129 814698129 IN IP4 *.*.*.*
s=Asterisk PBX 1.6.2.6
c=IN IP4 *.*.*.*
t=0 0
m=audio 10486 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

-- Called trunk-CloudSIP/0412345678

TE100*CLI>
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.21.0.240:5060;branch=z9hG4bK5836c158;rport=5060;received=10.21.0.240
From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as1407029f
To: <sip:0412345678@ipvoice.one:5060>
Call-ID: 0c46d23d4f8e683f4a10d1f74d327899@ipvoice.one
CSeq: 102 INVITE
Server: VProvider SIP proxy
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

TE100*CLI>
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.21.0.240:5060;received=10.21.0.240;branch=z9hG4bK5836c158;rport=5060
From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as1407029f
To: <sip:0412345678@ipvoice.one:5060>;tag=as3eb13934
Call-ID: 0c46d23d4f8e683f4a10d1f74d327899@ipvoice.one
CSeq: 102 INVITE
Server: VProvider
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0

 

Solution

1. Add/Edit the /persistent/etc/sip_extend.conf in the SSH.

change the parameter enableanonymous from "yes" to "no"

2. Fill the line number in the Caller ID field.

3. Reboot your gateway to let the change take effect. Then test again.

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