[SR-Users] remove_hf() leaves junk in the headers?

[Karsten Horsmann]

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_va

Cheers
Karsten Horsmann


Oded Arbel <odeda at 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 at 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 at example.com:5066>;tag=as0bcee217
> To: <sip:972522353283 at 10.20.30.40:5060>
> Contact: <sip:972515340014 at 10.90.80.71:5066>
> Call-ID: 722299dc15aad53d28172d433e8665f4 at 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 at 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 at 10.20.30.40:5060>
> Contact: <sip:972515340014 at 10.90.80.71:5066>
> Call-ID: 7b20ba5a549f6f886c22985146765736 at 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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