Incoming Caller ID only Numbers

Sangoma Trunking
1

I am setting up my first FreePBX system, so I am new to all this, but everything I read shows SIPStation should be forwarding Caller ID name headers. Debugging with sngrep, I am only seeing the phone number in the name fields. Is this something I’m doing wrong or an issue on the SIPStation side? What can I do to further debug?

I’m still in the trial for SIPStation, I have also filled out the form for a trial of ClearlyIP. Is there another provider I should try instead?

Below is my INVITE for the call (with my phone number and IP modified for privacy):

2025/03/08 10:40:19.601059 162.253.134.135:5060 -> 192.168.28.253:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 162.253.134.135;rport;branch=z9hG4bKUUKeBg23jr8Za
Max-Forwards: 69
From: "+19315551234" <sip:[email protected]>;tag=2NrcHeerKS73K
To: <sip:[email protected]:5060>
Call-ID: dd127c49-76de-123e-9e9e-00163c2074e3
CSeq: 96172217 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: SIPStation 2.11.3
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
X-A-Leg-Call-ID: [email protected]
X-FS-Support: update_display,send_info
P-Asserted-Identity: "+19315551234" <sip:[email protected]>

v=0
o=Sonus_UAC 571146 75748 IN IP4 67.231.13.111
s=SIP Media Capabilities
c=IN IP4 67.231.13.24
t=0 0
m=audio 21734 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
2

It’s not an Asterisk issue.

In your example, the name and number differ by the presence of a + in the name.

3

I was assuming this was a SIPStation issue, but was hoping for some tips to troubleshoot or verify.

4

It’s in the incoming requests:

P-Asserted-Identity: "+19315551234" <sip:[email protected]>
......................^^^^^^^^^^^^  Caller Name
.........................................^^^^^^^^^^^ Caller ID

Similarly for the From header, so definitely upstream of Asterisk.