Hi,
how can I remove/not send RPID and P-asserted identity?
Opensips sends to UAC (with RPID and p-asserted), uac sends back 302
request and that due to 302 I am doing new invite with opensips but in
this invite I can see RPID and p-asserted.
I am trying to remove it with remove_hf() but this does not works.
How can I deal with this issue.
here is a sip trace:
U opensips:5060 -> UAC_PUBLIC_IP:13647
INVITE sip:38618108753@UAC_PUBLIC_IP:13647 SIP/2.0.
Record-Route: <sip:opensips;lr;ftag=tarm9Ucep73Um;did=83e.cf63dc81>.
Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK96d7.e47216a1.0.
Via: SIP/2.0/UDP
RTP_IP:5080;received=RTP_IP;rport=5080;branch=z9hG4bK3X4y2N8F54H9a.
Max-Forwards: 67.
From: "38618108758" <sip:38618108758@RTP_IP>;tag=tarm9Ucep73Um.
To: <sip:38618108753@opensips>.
Call-ID: 5250f421-f243-1231-5695-005056b2fe3d.
CSeq: 54214780 INVITE.
Contact: <sip:mod_sofia@RTP_IP:5080>.
User-Agent:
FreeSWITCH-mod_sofia/1.2.17+git~20131230T193020Z~52377f0f65~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 207.
X-Call_id: 56c99a91-d0ef0551@172.31.1.103.
X-FS-Support: update_display,send_info.
P-Asserted-Identity: <sip:0038618108758@RTP_IP;user=phone>.
Remote-Party-ID: 0038618108758
<sip:0038618108758@RTP_IP>;party=calling;id-type=subscriber;privacy=off;screen=yes.
.
v=0.
o=FreeSWITCH 1389081737 1389081738 IN IP4 RTP_IP.
s=FreeSWITCH.
c=IN IP4 RTP_IP.
t=0 0.
m=audio 19952 RTP/AVP 0 8 9 101 13.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:30.
U UAC_PUBLIC_IP:13647 -> opensips:5060
SIP/2.0 302 Moved Temporarily.
To: <sip:38618108753@opensips>;tag=77cc99cb150aeeefi0.
From: "38618108758" <sip:38618108758@RTP_IP>;tag=tarm9Ucep73Um.
Call-ID: 5250f421-f243-1231-5695-005056b2fe3d.
CSeq: 54214780 INVITE.
Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK96d7.e47216a1.0.
Via: SIP/2.0/UDP
RTP_IP:5080;received=RTP_IP;rport=5080;branch=z9hG4bK3X4y2N8F54H9a.
Record-Route: <sip:opensips;lr;ftag=tarm9Ucep73Um;did=83e.cf63dc81>.
Contact: <sip:018108756@opensips>.
Diversion: "38618108753" <sip:38618108753@opensips>;reason=unconditional.
Server: Linksys/SPA922-6.1.5(a).
Content-Length: 0.
U opensips:5060 -> RTP_IP:5060
INVITE sip:30238618108756@opensips SIP/2.0.
Record-Route: <sip:opensips;lr;ftag=tarm9Ucep73Um;did=83e.cf63dc81>.
Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK96d7.e47216a1.1.
Via: SIP/2.0/UDP
RTP_IP:5080;received=RTP_IP;rport=5080;branch=z9hG4bK3X4y2N8F54H9a.
Max-Forwards: 67.
From: "38618108758" <sip:38618108758@RTP_IP>;tag=tarm9Ucep73Um.
To: <sip:38618108753@opensips>.
Call-ID: 5250f421-f243-1231-5695-005056b2fe3d.
CSeq: 54214780 INVITE.
Contact: <sip:mod_sofia@RTP_IP:5080>.
User-Agent:
FreeSWITCH-mod_sofia/1.2.17+git~20131230T193020Z~52377f0f65~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 207.
X-Call_id: 56c99a91-d0ef0551@172.31.1.103.
X-FS-Support: update_display,send_info.
P-Asserted-Identity: <sip:0038618108758@RTP_IP;user=phone>.
Remote-Party-ID: 0038618108758
<sip:0038618108758@RTP_IP>;party=calling;id-type=subscriber;privacy=off;screen=yes.
Moved: 38618108753.
.
v=0.
o=FreeSWITCH 1389081737 1389081738 IN IP4 RTP_IP.
s=FreeSWITCH.
c=IN IP4 RTP_IP.
t=0 0.
m=audio 19952 RTP/AVP 0 8 9 101 13.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:30.
tnx!
miha
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