Hi, Miha!
remove_hf() is the function you should use. According to your scenario,
you should remove the RPID and PAI in the failure route. Are you sure
that code is reached for the second INVITE?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 01/07/2014 03:52 PM, Miha wrote:
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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