[Freeswitch-users] Handling custom headers

[Cezary Siwek]

Hi all,

I'm trying to copy some of custom headers from A-leg to B-leg so I have 
tried this:

sip_h_P-HEADER1=${sip_h_P-HEADER1}

but it seems like I cannot access customer header variables ${sip_h_} if 
I don't set before sip_copy_custom_headers=true.
If I do that, all the headers are being copied automatically from LegA 
to LegB.
Is there any other way of doing it? I don't need them all to be copied. 
I know I can remove them afterwards but since they are dynamic I cannot 
predict how would they look like.


Another issue I have is that Freeswitch seems to be adding locally 
generated P-Asserted-Identity (based on From header) to  every final 
response. I tried to disable it by settiing sip_cid_type=none but no 
luck so far.  Removing it by 'unset' did not help either - PAI header is 
always there.

here is an example:

U 10.0.6.136:5060 -> 10.0.3.56:5060
INVITE sip:+4470000606 at example.domain.comSIP/2.0.
Via: SIP/2.0/UDP 10.0.6.136:5060;branch=z9hG4bK-3766-1-0.
From: <sip:+447000000155 at example.domain.com>;tag=1.
To: <sip:+4470000606 at example.domain.com:5060>.
Call-ID: ABC///1-3766 at 10.0.6.136.
CSeq: 1 INVITE.
Contact: sip:+447000000155 at 10.0.6.136:5060.
Max-Forwards: 10.
P-Asserted-Identity: sip:+447000000155 at example.domain.com.
Content-Type: application/sdp.
Content-Length:   133.
.
v=0.
o=user1 53655765 2353687637 IN IP4 10.0.6.136.
s=-.
c=IN IP4 10.0.6.136.
t=0 0.
m=audio 6000 RTP/AVP 97.
a=rtpmap:97 iLBC/8000



U 10.0.3.56:5060 -> 10.0.6.136:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.0.6.136:5060;branch=z9hG4bK-3766-1-0.
From: <sip:+447000000155 at example.domain.com>;tag=1.
To: <sip:+4470000606 at example.domain.com:5060>.
Call-ID: ABC///1-3766 at 10.0.6.136.
CSeq: 1 INVITE.
User-Agent: 
FreeSWITCH-mod_sofia/1.2.15+git~20131119T212207Z~a13c5005be~64bit.
Content-Length: 0.
.


U 10.0.3.56:5060 -> 10.0.6.136:5060
SIP/2.0 480 Temporarily Unavailable.
Via: SIP/2.0/UDP 10.0.6.136:5060;branch=z9hG4bK-3766-1-0.
Max-Forwards: 10.
From: <sip:+447000000155 at example.domain.com>;tag=1.
To: <sip:+4470000606 at example.domain.com:5060>;tag=2gK14HpU0cj9c.
Call-ID: ABC///1-3766 at 10.0.6.136.
CSeq: 1 INVITE.
User-Agent: 
FreeSWITCH-mod_sofia/1.2.15+git~20131119T212207Z~a13c5005be~64bit.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, 
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, conference, presence, dialog, line-seize, 
call-info, sla, include-session-description, presence.winfo, 
message-summary, refer.
Reason: Q.850;cause=19;text="NO_ANSWER".
Content-Length: 0.
P-Asserted-Identity: "+4470000606" <sip:+4470000606 at example.domain.com>.
.


U 10.0.6.136:5060 -> 10.0.3.56:5060
ACK sip:+4470000606 at example.domain.com:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.0.6.136:5060;branch=z9hG4bK-3766-1-0.
From: <sip:+447000000155 at example.domain.com>;tag=1.
To: <sip:+4470000606 at example.domain.com:5060>;tag=2gK14HpU0cj9c.
Call-ID: 1-3766 at 10.0.6.136.
CSeq: 1 ACK.
Contact: <sip:sipp at 10.0.6.136:5060;transport=UDP>.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.
.


I'm experiencing the same issue when trying to redirect with 302.

Regards
Cezary




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