I'm trying to rewrite the `From` header in an invite going out to an
external system (*), by calling remove_hf("From");
followed by
append_hf("From: <some value>\r\n");

but the result is an invalid SIP message, because apparently after
`remove_hf()` finishes, it leaves the content of $fU in place of the
`From` header - just the raw text, without a header name or even a
"\r\n" - and that eats the next header line and breaks the INVITE.

Here is how the INVITE originally looks (without either `remove_hf()`
or `append_hf()`):

---8<---
INVITE sip:972522353283@10.20.30.40:5060 SIP/2.0
Record-Route: <sip:10.90.80.70;lr;did=649.f0b>
Via: SIP/2.0/UDP
10.90.80.70:5060;branch=z9hG4bK6188.ff5c4fd855bd2970fdf32d4807b114c2.0
Via: SIP/2.0/UDP
10.90.80.71:5066;received=10.90.80.71;branch=z9hG4bK581897a4;rport=5066
Max-Forwards: 69
From: <sip:972515340014@example.com:5066>;tag=as0bcee217
To: <sip:972522353283@10.20.30.40:5060>
Contact: <sip:972515340014@10.90.80.71:5066>
Call-ID: 722299dc15aad53d28172d433e8665f4@10.90.80.71:5066
CSeq: 102 INVITE
User-Agent: Asterisk PBX 14.7.2
Date: Sun, 26 May 2019 14:42:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces
X-CID: ad199638-4df3-4aef-81f1-1b50581032a7
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 817636675 817636675 IN IP4 10.50.60.70
s=Asterisk PBX 14.7.2
c=IN IP4 10.50.60.70
t=0 0
m=audio 39064 RTP/AVP 8 0 18 9
a=maxptime:60
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtcp:39065
a=ptime:20
---8<---

and here is another call with the same setup, with just
`remove_hf("From");` being invoked (without calling `append_hf()` -
this is just for reproduction):

---8<---
INVITE sip:972522353283@10.20.30.40:5060 SIP/2.0
Record-Route: <sip:10.90.80.70;lr;did=f45.91b1>
Via: SIP/2.0/UDP
10.90.80.70:5060;branch=z9hG4bK024d.2690ebf7a809edca2208e6b2561ccf9e.0
Via: SIP/2.0/UDP
10.90.80.71:5066;received=10.90.80.71;branch=z9hG4bK79ae3073;rport=5066
Max-Forwards: 69
972515340014To: <sip:972522353283@10.20.30.40:5060>
Contact: <sip:972515340014@10.90.80.71:5066>
Call-ID: 7b20ba5a549f6f886c22985146765736@10.90.80.71:5066
CSeq: 102 INVITE
User-Agent: Asterisk PBX 14.7.2
Date: Sun, 26 May 2019 14:44:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces
X-CID: ffbb4ac1-3a0f-4dd6-9ce9-7ad24361081a
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 509740923 509740923 IN IP4 10.50.60.70
s=Asterisk PBX 14.7.2
c=IN IP4 10.50.60.70
t=0 0
m=audio 39102 RTP/AVP 8 0 18 9
a=maxptime:60
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtcp:39103
a=ptime:20
---8<---

As you can see, the `From` header is indeed gone, but it left the $fU
just before the `To` header.

WTF?

I'm running Kamailio 5.2.2 built locally.

Notes:
*) the problem I'm trying to solve is that I can't get my Asterisk
installation to generate the `From` header I want, the reason is out
of scope of this question, so I'm sending custom SIP headers over to
Kamaiio and I want to rewrite the header in Kamailio to the "correct"
value.

--
Oded Arbel

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