Hi Oded,
I would better use uac_replace_from like described in the Kamailio wiki. (link below) This function takes also care of rewriting it back AFAIK.
http://www.kamailio.org/wiki/tutorials/faq/main#how_to_set_different_header_...
Cheers Karsten Horsmann
Oded Arbel odeda@cloudonix.io schrieb am So., 26. Mai 2019, 17:02:
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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