In ngrep traffic check no active rdp-session-id
but do not know how to solve
#
U +3.135110 IP-ASTERISK:5060 -> IP_OPENSIPS:5060
INVITE sip:100@ IP_OPENSIPS SIP/2.0
Via: SIP/2.0/UDP IP-ASTERISK:5060;branch=z9hG4bK3e684698;rport
Max-Forwards: 70
From: "3414741468"
<sip:TRK00253-001@IP-ASTERISK>;tag=as33306c2a
To: <sip:100@IP_OPENSIPS>
Contact: <sip:TRK00253-001@IP-ASTERISK>
Call-ID: 46ea6e9819e3583c59479d9304cc2b4f@IP-ASTERISK
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Mon, 26 Mar 2012 16:29:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 1324806659 1324806659 IN IP4 IP-ASTERISK
s=Asterisk PBX 1.6.2.20
c=IN IP4 IP-ASTERISK
t=0 0
m=audio 10788 RTP/AVP 0 18 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
tanks
Bogdan-Andrei Iancu wrote:
Well, you know, one is what we want to do , another we actually
get.
I was rather asking if, making a sip capture (with ngrep) you see
in your call the RTPproxy insertion - check it in traffic, not in
script.
Regards,
Bogdan
On 04/02/2012 10:05 PM, magnusadil...@gmail.com
wrote:
hi, yes, rtpproxy is active in invite 200
onreply_route[3] {
if ((isflagset(5) || isbflagset(0)) && status =~
"(183)|(2[0-9][0-9])" && has_body("application/sdp")) {
if (rtpproxy_answer()) {
log("L_INFO: rtpproxy_answer NAT");
}
}
if (!subst_uri('/(sip:.*);nat=yes/\1/'))
{
search_append('Contact:.*sip:[^>[:cntrl:]]*',
';nat=yes');
}
exit;
}
But i'm implemented this in invite route
if (is_method("INVITE") {
if ($si == "IP ASTERISK" && is_method("INVITE")) {
fix_nated_contact();
fix_nated_sdp("1");
xlog("L_INFO", "NAT detected3 PSTN for SIP");
setflag(5);
return;
}
}
and worked, but
I think it is not correct
tansk
Bogdan-Andrei Iancu wrote:
Hi Magnus,
attaching cfg files is useless, as no one will debug the
script, but we will help you to debug your script.
So, for the non-working case (PSTN to SIP) does your script
force RTPproxy in INVITE and 200 OK ?
Regards,
Bogdan
On 03/29/2012 01:52 AM, magnusadil...@gmail.com
wrote:
I have phones (some behind NAT) connecting to Opensips
server an Asterisk and an rtpproxy as seen below:
rtpproxy started with
ps -aux | grep rtpproxy
root 15666 0.0 0.0 14472 920 ? Ssl Mar23
0:05 ./rtpproxy -F -l 189.254.2.19 -s udp:* 7890 -d DBUG
LOG_LOCAL3
UAC1 username =
100------------Firewall/router--------------------Opensips
1.7---------- RTP PROXY------------Asterisk 1.6
192.168.1.10
192.168.1.1 65.254.63.212
189.254.2.19 190.61.201.89
external ip dinamic 169.254.2.2
- Calls between UAC are OK (both SIP and RTP).
- Calls UAC for PSTN is OK.
- Did numbers is received in Asterisk, and destination for
UAC registered in opensips, but no work audio .
(EX User call cellphone for DID 54115368566, call is
received in asterisk, and destination for user 100,
registered in opensips)
_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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