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The relevant one should be INVITE leaving you opensips - to see if
RTPproxy was inserted in the SDP.<br>
<br>
Regards,<br>
Bogdan<br>
<br>
On 04/02/2012 11:41 PM, <a class="moz-txt-link-abbreviated" href="mailto:magnusadilsom@gmail.com">magnusadilsom@gmail.com</a> wrote:
<blockquote cite="mid:4F7A0EF1.6040403@gmail.com" type="cite">
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
In ngrep traffic check no active rdp-session-id<br>
<br>
but do not know how to solve<br>
<br>
<br>
#<br>
U +3.135110 IP-ASTERISK:5060 -> IP_OPENSIPS:5060<br>
INVITE <a moz-do-not-send="true" class="moz-txt-link-freetext"
href="sip:100@">sip:100@</a> IP_OPENSIPS SIP/2.0 <br>
Via: SIP/2.0/UDP IP-ASTERISK:5060;branch=z9hG4bK3e684698;rport <br>
Max-Forwards: 70 <br>
From: "3414741468" <a moz-do-not-send="true"
class="moz-txt-link-rfc2396E"
href="sip:TRK00253-001@IP-ASTERISK"><sip:TRK00253-001@IP-ASTERISK></a>;tag=as33306c2a
<br>
To: <a moz-do-not-send="true" class="moz-txt-link-rfc2396E"
href="sip:100@IP_OPENSIPS"><sip:100@IP_OPENSIPS></a> <br>
Contact: <a moz-do-not-send="true" class="moz-txt-link-rfc2396E"
href="sip:TRK00253-001@IP-ASTERISK"><sip:TRK00253-001@IP-ASTERISK></a>
<br>
Call-ID: 46ea6e9819e3583c59479d9304cc2b4f@IP-ASTERISK<br>
CSeq: 102 INVITE <br>
User-Agent: Asterisk PBX 1.6.2.20 <br>
Date: Mon, 26 Mar 2012 16:29:17 GMT <br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO <br>
Supported: replaces, timer <br>
Content-Type: application/sdp <br>
Content-Length: 333 <br>
<br>
v=0 <br>
o=root 1324806659 1324806659 IN IP4 IP-ASTERISK <br>
s=Asterisk PBX 1.6.2.20 <br>
c=IN IP4 IP-ASTERISK<br>
t=0 0 <br>
m=audio 10788 RTP/AVP 0 18 8 3 101 <br>
a=rtpmap:0 PCMU/8000 <br>
a=rtpmap:18 G729/8000 <br>
a=fmtp:18 annexb=no <br>
a=rtpmap:8 PCMA/8000 <br>
a=rtpmap:3 GSM/8000 <br>
a=rtpmap:101 telephone-event/8000 <br>
a=fmtp:101 0-16 <br>
a=ptime:20 <br>
a=sendrecv <br>
<br>
<br>
<br>
tanks <br>
<br>
<br>
<br>
<br>
<br>
Bogdan-Andrei Iancu wrote:
<blockquote cite="mid:4F7A088D.6040807@opensips.org" type="cite">
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
Well, you know, one is what we want to do , another we actually
get.<br>
<br>
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.<br>
<br>
Regards,<br>
Bogdan<br>
<br>
On 04/02/2012 10:05 PM, <a moz-do-not-send="true"
class="moz-txt-link-abbreviated"
href="mailto:magnusadilsom@gmail.com">magnusadilsom@gmail.com</a>
wrote:
<blockquote cite="mid:4F79F882.7060407@gmail.com" type="cite">
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
hi, yes, rtpproxy is active in invite 200<br>
<br>
onreply_route[3] {<br>
if ((isflagset(5) || isbflagset(0)) && status =~
"(183)|(2[0-9][0-9])" && has_body("application/sdp"))
{<br>
if (rtpproxy_answer()) {<br>
log("L_INFO: rtpproxy_answer NAT");<br>
}<br>
}<br>
if (!subst_uri('/(<a moz-do-not-send="true"
class="moz-txt-link-freetext" href="sip:.*">sip:.*</a>);nat=yes/\1/'))
{<br>
search_append('Contact:.*sip:[^>[:cntrl:]]*',
';nat=yes');<br>
}<br>
exit;<br>
}<br>
<br>
<br>
But i'm implemented this in invite route<br>
<br>
if (is_method("INVITE") {<br>
if ($si == "IP ASTERISK" && is_method("INVITE"))
{<br>
fix_nated_contact();<br>
fix_nated_sdp("1");<br>
xlog("L_INFO", "NAT detected3 PSTN for SIP");<br>
setflag(5);<br>
return;<br>
}<br>
}<br>
<br>
<span id="result_box" class="short_text" lang="en"><span
class="hps">and</span> <span class="hps">worked, </span></span><span
id="result_box" class="short_text" lang="en"><span
class="hps">but I think it</span> <span class="hps">is
not</span> <span class="hps">correct</span></span><br>
<br>
tansk<br>
<br>
<br>
Bogdan-Andrei Iancu wrote:
<blockquote cite="mid:4F79D798.1080707@opensips.org"
type="cite">
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
Hi Magnus,<br>
<br>
attaching cfg files is useless, as no one will debug the
script, but we will help you to debug your script.<br>
<br>
So, for the non-working case (PSTN to SIP) does your script
force RTPproxy in INVITE and 200 OK ?<br>
<br>
Regards,<br>
Bogdan<br>
<br>
On 03/29/2012 01:52 AM, <a moz-do-not-send="true"
class="moz-txt-link-abbreviated"
href="mailto:magnusadilsom@gmail.com">magnusadilsom@gmail.com</a>
wrote:
<blockquote cite="mid:4F739644.30100@gmail.com" type="cite">
<meta http-equiv="content-type" content="text/html;
charset=ISO-8859-1">
I have phones (some behind NAT) connecting to Opensips
server an Asterisk and an rtpproxy as seen below:<br>
<br>
rtpproxy started with<br>
ps -aux | grep rtpproxy<br>
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 <br>
<br>
<br>
<br>
UAC1 username =
100------------Firewall/router--------------------Opensips
1.7---------- RTP PROXY------------Asterisk 1.6<br>
192.168.1.10
192.168.1.1 65.254.63.212
189.254.2.19 190.61.201.89<br>
external ip dinamic 169.254.2.2<br>
<br>
<br>
- Calls between UAC are OK (both SIP and RTP).<br>
- Calls UAC for PSTN is OK.<br>
- Did numbers is received in Asterisk, and destination for
UAC registered in opensips, but no work audio .<br>
(EX User call cellphone for DID 54115368566, call is
received in asterisk, and destination for user 100,
registered in opensips)<br>
</blockquote>
</blockquote>
</blockquote>
</blockquote>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
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