How to hide outbound CID for extension

jkalber · April 8, 2025, 4:52pm

Hello, I have an employee at an office that would like to have the option to place calls outbound anonymously. We are located in the USA. I figured it was as easy as having him input *67 before the phone number but it isn’t working.

When he dials *67 and then a phone number and attempts to place a call, there is a message on his Yealink T42G phone that says “Not found!” and then the call ends.

I hopped on our FreePBX Admin Portal, navigated to his extension but I’m not seeing any options there to hide his CID/Number for outbound calls.

david55 · April 8, 2025, 5:16pm

How are you connected to the PSTN? If via SIP, what are the provider’s rules for caller ID, and for the use of the From user field?

jkalber · April 8, 2025, 5:31pm

Our SIP trunk provider is Vitelity (an Inteliquent Company) and I am unsure on what their rules are for caller ID, and for the use of the From user field. So you are saying I need to first reach out to Vitelity to see if anonymous calling is even an option?

jkalber · April 8, 2025, 5:44pm

I opened a ticket with Vitelity and this was their response below. So it does look like I have the ability to place anonymous calls. I’m just not seeing anywhere in the FreePBX Admin Portal to make that change for this specific employee. Any help is greatly appreciated.

“You are welcome to obfuscate your From Number/Caller ID as you see fit. We will simple pass along your From Number as you send it to us.
However, the trouble you will run into with this is that most of the carriers that you call have every right to reject your call, and most will do so. We would have no recourse for this. The only work-around would be to send a proper 10-digit number instead.”

adell4444 · April 8, 2025, 6:10pm

Create an outbound route with a dial pattern that requires a prefix (e.g. 67). Then, override the caller ID on that outbound route with “anonymous”.

tmittelstaedt · April 8, 2025, 6:58pm

The company I work for - medical provider - has users who demanded this on the theory that calls from the medical center going to a landline leak patient information. We implemented this a decade ago and things were fine for a few years but now, we are getting tons of complaints from them that patients are all using cell phones and blocking their calls. So we put in a *9 for them to reveal the company phone number and now they are complaining that they forget to dial that in.

My advice is you tell the employee once you implement this they have no right to complain to you when they dial *67 and can’t get through.

jkalber · April 8, 2025, 8:00pm

I did just as you suggested but still encounter the same issue. When I dial *67 plus the phone number and then click Send on my desk phone to place the call, there are 3 busy tone beeps and then the call ends. On the phone display itself it reads “Call Failed! Not Found”.

david55 · April 8, 2025, 8:04pm

You are matching 67, not *67

jkalber · April 8, 2025, 8:12pm

I put a * in front and saved my changes. Just tried to place an anonymous call. The three beeps/busy tone no longer happens. Now it says “All circuits are busy, please try your call again later”. Then the phone call ends.

jkalber · April 9, 2025, 7:28pm

I was hoping that maybe the PBX system needed a reboot, did that but still facing the same issue unfortunately.

david55 · April 9, 2025, 8:11pm

Please provide /var/log/asterisk/full, enabled if necessary, and with verbosity at least 5.

DarrellSpears · April 12, 2025, 3:42pm

The key word “hidden” is what I believe you should be using.

jkalber · April 14, 2025, 5:07pm

Ok thank you, I’ll give that a try tomorrow when I am back in the office.

DarrellSpears · April 14, 2025, 5:53pm

I believe that should work for you providing your VoIP provider allows anonymous caller id. I’ve been using this for over 10 years and I tested it yesterday and it works for me. The word hidden blocks the name and number. You don’t have to code anonymous. The telephone company provides that detail when they see the caller id is hidden. Some telephone companies may display “OUT OF AREA” for hidden caller id. It depends on the phone company that your call is routed to.

sam · April 14, 2025, 6:55pm

I’m also trying to hide my caller ID, and I did as you said here. I’ve tried in the past as well but all I’ve ever gotten was an “all circuits are busy now” error when I call out. My VoIP provider is Telnyx. I’ve messed with the settings below but still never had any luck. From my previous experiences, I think I had to do something with a custom SIP header. I’m not sure. I’d appreciate any help.

After turning on pjsip logging, I got this error:

I tried what this post suggested, but it still didn’t work.

david55 · April 14, 2025, 7:28pm

I don’t think “hidden” gets sent over the wire. What actually gets sent is what you see in the image (@sam please use the actual plain text, from the log, not a image, or rekeyed text).

david55 · April 14, 2025, 7:32pm

I suspect the provider requires the real caller ID in the From header (probably involves setting from user), and that Send RPID/PAI be something other than No. It’s possible they simply don’t allow anonymous calls. You should ask them what they require.

sam · April 14, 2025, 7:50pm

You’re right, Telnyx says:

You are still required to send a valid origination number with the Privacy header. We will in turn change the caller ID to anonymous. However if we do not receive a valid caller ID the calls will be rejected with the error 403 Caller Origination Number is Invalid D35.

I did receive that D35 error now that I looked. Right now, I have the extension display name as just sam with no outbound CID. For the route, I have a prefix of *67 set up so when I dial with that, it selects this route. It has a Route CID of hidden. Override extension is off. For the trunk, I have the outbound CID as the actual Telnyx number with Hide CID off and Allow Any CID checked. I also have the trunk sending both RPID and PAI, with sending Private Caller ID Information enabled. I receive an “all circuits are busy” error and still receive the D35 error in the console. How can I configure FreePBX to send a valid From header while also having it be an anonymous call? Is it something with the settings I listed here? Thanks!

david55 · April 14, 2025, 8:54pm

Your log shows neither RPID nor PAI. Is it complete?

sam · April 14, 2025, 9:10pm

Here is the complete log from start to finish:

PJSIP Log

<--- Received SIP request (965 bytes) from UDP:192.168.1.9:57696 --->
INVITE sip:*[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:57696;branch=z9hG4bK-524287-1---bb55f86204b1da18;rport
Max-Forwards: 70
Contact: <sip:[email protected]:57696;transport=UDP>
To: <sip:*[email protected]>
From: <sip:[email protected];transport=UDP>;tag=d1f49463
Call-ID: OdyPgNfRUnG2Bq4aMT-_aA..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.4_1
Allow-Events: presence, kpml, talk, as-feature-event
X-Push-Stop: yes
Content-Length: 258

v=0
o=Z 0 680863418 IN IP4 192.168.1.9
s=Z
c=IN IP4 192.168.1.9
t=0 0
m=audio 64038 RTP/AVP 3 101 110 97 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP response (511 bytes) to UDP:192.168.1.9:57696 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.9:57696;rport=57696;received=192.168.1.9;branch=z9hG4bK-524287-1---bb55f86204b1da18
Call-ID: OdyPgNfRUnG2Bq4aMT-_aA..
From: <sip:[email protected]>;tag=d1f49463
To: <sip:*[email protected]>;tag=z9hG4bK-524287-1---bb55f86204b1da18
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1744664271/8e82cdcaf07a5616971415ed1d923962",opaque="58ed608954b60ccc",algorithm=MD5,qop="auth"
Server: FPBX-17.0.19.24(21.7.0)
Content-Length:  0

<--- Received SIP request (365 bytes) from UDP:192.168.1.9:57696 --->
ACK sip:*[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:57696;branch=z9hG4bK-524287-1---bb55f86204b1da18;rport
Max-Forwards: 70
To: <sip:*[email protected]>;tag=z9hG4bK-524287-1---bb55f86204b1da18
From: <sip:[email protected];transport=UDP>;tag=d1f49463
Call-ID: OdyPgNfRUnG2Bq4aMT-_aA..
CSeq: 1 ACK
Content-Length: 0

<--- Received SIP request (1270 bytes) from UDP:192.168.1.9:57696 --->
INVITE sip:*[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:57696;branch=z9hG4bK-524287-1---1487892c34d2c211;rport
Max-Forwards: 70
Contact: <sip:[email protected]:57696;transport=UDP>
To: <sip:*[email protected]>
From: <sip:[email protected];transport=UDP>;tag=d1f49463
Call-ID: OdyPgNfRUnG2Bq4aMT-_aA..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.4_1
Authorization: Digest username="600",realm="asterisk",nonce="1744664271/8e82cdcaf07a5616971415ed1d923962",uri="sip:*[email protected];transport=UDP",response="37183f95240ad1cddc34365973f72a8a",cnonce="f688d12cba8c679a42c2568abf96ad7d",nc=00000001,qop=auth,algorithm=MD5,opaque="58ed608954b60ccc"
Allow-Events: presence, kpml, talk, as-feature-event
X-Push-Stop: yes
Content-Length: 258

v=0
o=Z 0 680863418 IN IP4 192.168.1.9
s=Z
c=IN IP4 192.168.1.9
t=0 0
m=audio 64038 RTP/AVP 3 101 110 97 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP response (319 bytes) to UDP:192.168.1.9:57696 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.9:57696;rport=57696;received=192.168.1.9;branch=z9hG4bK-524287-1---1487892c34d2c211
Call-ID: OdyPgNfRUnG2Bq4aMT-_aA..
From: <sip:[email protected]>;tag=d1f49463
To: <sip:*[email protected]>
CSeq: 2 INVITE
Server: FPBX-17.0.19.24(21.7.0)
Content-Length:  0

[2025-04-14 16:57:51] ERROR[144728]: res_pjsip_header_funcs.c:726 remove_header: No headers had been previously added to this session.
<--- Transmitting SIP request (1192 bytes) to UDP:192.76.120.10:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XXX.XX.XX:5060;rport;branch=z9hG4bKPj9dc7a56e-7764-4728-9b2e-cd35fbdcd81b
From: "Anonymous" <sip:[email protected]>;tag=0df42ac2-aacf-4937-b587-46fd367bbe7c
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 242c9f02-886f-4754-b5c1-0b636c522e2b
CSeq: 23327 INVITE
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Privacy: id
P-Asserted-Identity: "hidden" <sip:192.168.1.24>
Remote-Party-ID: "hidden" <sip:192.168.1.24>;party=calling;privacy=full;screen=yes
Max-Forwards: 70
User-Agent: FPBX-17.0.19.24(21.7.0)
Content-Type: application/sdp
Content-Length:   337

v=0
o=- 859798138 859798138 IN IP4 XX.XXX.XX.XX
s=Asterisk
c=IN IP4 XX.XXX.XX.XX
t=0 0
m=audio 19966 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (394 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 100 Telnyx Trying
Via: SIP/2.0/UDP 192.168.1.24:5060;rport=5060;received=192.168.1.24;branch=z9hG4bKPj9dc7a56e-7764-4728-9b2e-cd35fbdcd81b
From: "Anonymous" <sip:[email protected]>;tag=0df42ac2-aacf-4937-b587-46fd367bbe7c
To: <sip:[email protected]>
Call-ID: 242c9f02-886f-4754-b5c1-0b636c522e2b
CSeq: 23327 INVITE
Server: Telnyx SIP Proxy
Content-Length: 0

<--- Received SIP response (752 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.24:5060;received=192.168.1.24;rport=5060;branch=z9hG4bKPj9dc7a56e-7764-4728-9b2e-cd35fbdcd81b
From: "Anonymous" <sip:[email protected]>;tag=0df42ac2-aacf-4937-b587-46fd367bbe7c
To: <sip:[email protected]>;tag=vSyBBBQeS2emc
Call-ID: 242c9f02-886f-4754-b5c1-0b636c522e2b
CSeq: 23327 INVITE
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
Supported: path
Allow-Events: talk, hold, conference, refer
Proxy-Authenticate: Digest realm="sip.telnyx.com", nonce="7e86afa3-cbb9-4b5b-9eea-41a657f04353", algorithm=MD5, qop="auth", opaque="f959351f-b59f-4849-a61a-4e9b16c833c1/10.13.212.4"
Content-Length: 0

<--- Transmitting SIP request (433 bytes) to UDP:192.76.120.10:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XXX.XX.XX:5060;rport;branch=z9hG4bKPj9dc7a56e-7764-4728-9b2e-cd35fbdcd81b
From: "Anonymous" <sip:[email protected]>;tag=0df42ac2-aacf-4937-b587-46fd367bbe7c
To: <sip:[email protected]>;tag=vSyBBBQeS2emc
Call-ID: 242c9f02-886f-4754-b5c1-0b636c522e2b
CSeq: 23327 ACK
Max-Forwards: 70
User-Agent: FPBX-17.0.19.24(21.7.0)
Content-Length:  0

<--- Transmitting SIP request (1549 bytes) to UDP:192.76.120.10:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XXX.XX.XX:5060;rport;branch=z9hG4bKPjca9a4b34-8d46-476b-8bb4-e94d698ac7a3
From: "Anonymous" <sip:[email protected]>;tag=0df42ac2-aacf-4937-b587-46fd367bbe7c
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 242c9f02-886f-4754-b5c1-0b636c522e2b
CSeq: 23328 INVITE
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Privacy: id
Max-Forwards: 70
User-Agent: FPBX-17.0.19.24(21.7.0)
Proxy-Authorization: Digest username="XXXXXXXXXXXXXX", realm="sip.telnyx.com", nonce="7e86afa3-cbb9-4b5b-9eea-41a657f04353", uri="sip:[email protected]:5060", response="920204203520079295905521e4e25611", algorithm=MD5, cnonce="5f34cd178fc04339a0a722f3d8a6d468", opaque="f959351f-b59f-4849-a61a-4e9b16c833c1/10.13.212.4", qop=auth, nc=00000001
P-Asserted-Identity: "hidden" <sip:192.168.1.24>
Remote-Party-ID: "hidden" <sip:192.168.1.24>;party=calling;privacy=full;screen=yes
Content-Type: application/sdp
Content-Length:   337

v=0
o=- 859798138 859798138 IN IP4 XX.XXX.XX.XX
s=Asterisk
c=IN IP4 XX.XXX.XX.XX
t=0 0
m=audio 19966 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (394 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 100 Telnyx Trying
Via: SIP/2.0/UDP 192.168.1.24:5060;rport=5060;received=192.168.1.24;branch=z9hG4bKPjca9a4b34-8d46-476b-8bb4-e94d698ac7a3
From: "Anonymous" <sip:[email protected]>;tag=0df42ac2-aacf-4937-b587-46fd367bbe7c
To: <sip:[email protected]>
Call-ID: 242c9f02-886f-4754-b5c1-0b636c522e2b
CSeq: 23328 INVITE
Server: Telnyx SIP Proxy
Content-Length: 0

<--- Received SIP response (643 bytes) from UDP:192.76.120.10:5060 --->
SIP/2.0 403 Caller Origination Number is Invalid D35
Via: SIP/2.0/UDP 192.168.1.24:5060;received=192.168.1.24;rport=5060;branch=z9hG4bKPjca9a4b34-8d46-476b-8bb4-e94d698ac7a3
Max-Forwards: 69
From: "Anonymous" <sip:[email protected]>;tag=0df42ac2-aacf-4937-b587-46fd367bbe7c
To: <sip:[email protected]>;tag=X2Q4c67HpB56Q
Call-ID: 242c9f02-886f-4754-b5c1-0b636c522e2b
CSeq: 23328 INVITE
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
Supported: path
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=21;text="CALL_REJECTED"
Content-Length: 0

<--- Transmitting SIP request (433 bytes) to UDP:192.76.120.10:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XXX.XX.XX:5060;rport;branch=z9hG4bKPjca9a4b34-8d46-476b-8bb4-e94d698ac7a3
From: "Anonymous" <sip:[email protected]>;tag=0df42ac2-aacf-4937-b587-46fd367bbe7c
To: <sip:[email protected]>;tag=X2Q4c67HpB56Q
Call-ID: 242c9f02-886f-4754-b5c1-0b636c522e2b
CSeq: 23328 ACK
Max-Forwards: 70
User-Agent: FPBX-17.0.19.24(21.7.0)
Content-Length:  0

<--- Transmitting SIP response (896 bytes) to UDP:192.168.1.9:57696 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.9:57696;rport=57696;received=192.168.1.9;branch=z9hG4bK-524287-1---1487892c34d2c211
Call-ID: OdyPgNfRUnG2Bq4aMT-_aA..
From: <sip:[email protected]>;tag=d1f49463
To: <sip:*[email protected]>;tag=436e5356-c6b7-4b63-87e0-1fc4ec6bae28
CSeq: 2 INVITE
Server: FPBX-17.0.19.24(21.7.0)
Contact: <sip:192.168.1.24:5060>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
P-Asserted-Identity: "CID:(Hidden)" <sip:[email protected]>
Content-Type: application/sdp
Content-Length:   274

v=0
o=- 0 680863420 IN IP4 192.168.1.24
s=Asterisk
c=IN IP4 192.168.1.24
t=0 0
m=audio 12344 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP request (687 bytes) from UDP:192.168.1.9:57696 --->
CANCEL sip:*[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:57696;branch=z9hG4bK-524287-1---1487892c34d2c211;rport
Max-Forwards: 70
To: <sip:*[email protected]>
From: <sip:[email protected];transport=UDP>;tag=d1f49463
Call-ID: OdyPgNfRUnG2Bq4aMT-_aA..
CSeq: 2 CANCEL
User-Agent: Zoiper v2.10.20.4_1
Authorization: Digest username="600",realm="asterisk",nonce="1744664271/8e82cdcaf07a5616971415ed1d923962",uri="sip:*[email protected];transport=UDP",response="1fb17e5171cd837da298f7c4e9432f6a",cnonce="e0c2664d563d7669e2cd114dabe0a32c",nc=00000002,qop=auth,algorithm=MD5,opaque="58ed608954b60ccc"
X-Push-Stop: yes
Content-Length: 0

<--- Transmitting SIP response (356 bytes) to UDP:192.168.1.9:57696 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:57696;rport=57696;received=192.168.1.9;branch=z9hG4bK-524287-1---1487892c34d2c211
Call-ID: OdyPgNfRUnG2Bq4aMT-_aA..
From: <sip:[email protected]>;tag=d1f49463
To: <sip:*[email protected]>;tag=436e5356-c6b7-4b63-87e0-1fc4ec6bae28
CSeq: 2 CANCEL
Server: FPBX-17.0.19.24(21.7.0)
Content-Length:  0

<--- Transmitting SIP response (556 bytes) to UDP:192.168.1.9:57696 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.9:57696;rport=57696;received=192.168.1.9;branch=z9hG4bK-524287-1---1487892c34d2c211
Call-ID: OdyPgNfRUnG2Bq4aMT-_aA..
From: <sip:[email protected]>;tag=d1f49463
To: <sip:*[email protected]>;tag=436e5356-c6b7-4b63-87e0-1fc4ec6bae28
CSeq: 2 INVITE
Server: FPBX-17.0.19.24(21.7.0)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
P-Asserted-Identity: "CID:(Hidden)" <sip:[email protected]>
Content-Length:  0

<--- Received SIP request (366 bytes) from UDP:192.168.1.9:57696 --->
ACK sip:*[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:57696;branch=z9hG4bK-524287-1---1487892c34d2c211;rport
Max-Forwards: 70
To: <sip:*[email protected]>;tag=436e5356-c6b7-4b63-87e0-1fc4ec6bae28
From: <sip:[email protected];transport=UDP>;tag=d1f49463
Call-ID: OdyPgNfRUnG2Bq4aMT-_aA..
CSeq: 2 ACK
Content-Length: 0```