Hello,
I have a PJSIP trunk with voip.ms and I’m trying to set up outbound caller ID blocking. I’ve set up an outbound route matching calls starting with *67, and configured the caller ID on this route as “hidden” using my valid DID number. The problem is that voip.ms blocks these calls because the caller ID number is not sent, which voip.ms requires. I contacted their support, and they indicated that I must pass a valid Caller ID number and set the “privacy=on” flag in the Remote-Party-ID SIP header to instruct voip.ms to hide my number. I’m not sure how to do this. I enabled Send RPID/PAI in the PJSIP trunk options, but this didn’t help.
Using tcpdump, I’ve confirmed that when I set “hidden” in the Caller ID field, FreePBX does not send the Remote-Party-ID field even if it’s enabled in the trunk PJSIP options:
raspbx.home.lan.sip > <voipms-server>.sip: [udp sum ok] SIP, length: 941
INVITE sip:<number>@<server>:5060 SIP/2.0
Via: SIP/2.0/UDP <my-IP>:5060;rport;branch=z9hG4bKPj7911f74a-e1db-471f-991f-75261e935ffb
From: "Anonymous" <sip:[email protected]>;tag=52ac39b5-4095-4e6e-a7c2-2d8f39859a2e
To: <sip:<number>@<server>>
Contact: <sip:asterisk@<my-IP>:5060>
Call-ID: <call-ID>
CSeq: 699 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-15.0.16.49(16.9.0)
Content-Type: application/sdp
Content-Length: 241
v=0
o=- 1897907294 1897907294 IN IP4 <my-IP>
s=Asterisk
c=IN IP4 <my-IP>
t=0 0
m=audio 19480 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
However, if I don’t set “hidden” in the Caller ID field, then the Remote-Party-ID is added as expected but Privacy is set to off, not on. voip.ms needs for it to be ON to hide the caller ID, and I don’t know how to do this.
22:14:57.742357 IP (tos 0x60, ttl 64, id 8587, offset 0, flags [DF], proto UDP (17), length 1050)
raspbx.home.lan.sip > <voipms-server>.sip: [udp sum ok] SIP, length: 1022
INVITE sip:<number>@<voipms-server>:5060 SIP/2.0
Via: SIP/2.0/UDP <my-IP>:5060;rport;branch=z9hG4bKPjfef2c145-ace0-44e2-9c7d-4dff06e43841
From: "test" <sip:<number>@<my-IP>>;tag=8dbc786f-8138-404a-ae0e-5dbb76a9f421
To: <sip:<number>@<voipms-server>>
Contact: <sip:asterisk@<my-ip>:5060>
Call-ID: <caller-ID string>
CSeq: 27676 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: "test" <sip:<number>@<IP>>;party=calling;privacy=off;screen=yes
Max-Forwards: 70
User-Agent: FPBX-15.0.16.49(16.9.0)
Content-Type: application/sdp
Content-Length: 239
v=0
o=- 178299335 178299335 IN IP4 <my-IP>
s=Asterisk
c=IN IP4 <my-IP>
t=0 0
m=audio 15588 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
Thanks
cinergi