<div dir="ltr">Take a look over this howto <br><br><a href="http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English">http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English</a><br>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/9/28 Mike Tesliuk <span dir="ltr"><<a href="mailto:mike@ultra.net.br" target="_blank">mike@ultra.net.br</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div><div><div><div><div><div><div><div><div><br></div><div>---- if i say something wrong please somebody correct my ----<br></div><div><br><br>Hello Rajesh, <br><br></div>You are using the nat_uac_test with parameter 23, this means parameters 16, 4, 2, 1 , what means <br>
<br><ul type="disc"><li><p>
<span><em>1</em></span> - Contact header field is searched
for occurrence of RFC1918 addresses.
</p></li><li><p>
<span><em>2</em></span> - the "received" test is used: address
in Via is compared against source IP address of signaling
</p></li><li><p>
<span><em>4</em></span> - Top Most VIA is searched
for occurrence of RFC1918 addresses
</p></li><li><p>
<span><em>16</em></span> - test if the source port is different
from the port in Via
</p></li></ul><br><br></div>i dont know if you understand but this is a binary count, you can check in this way<br><br></div>0010111 -> this is what you turn on<br><br></div>in this case, if your package does not contains an Private ip address on contact header, or does not contains a received on VIA different from the ip address of the signalling, does not contais on VIA an private ip address and the source port is not different from port on VIA , so your rule will not match (just on match is enought) <br>
<br></div>Look at this invite below (sended from a zoiper)<br><br> <a href="http://204.16.0.26:60340" target="_blank">204.16.0.26:60340</a> -> <a href="http://204.16.1.50:5060" target="_blank">204.16.1.50:5060</a><br>
INVITE <a href="mailto:sip%3A101@204.16.1.50" target="_blank">sip:101@204.16.1.50</a>;transport=UDP SIP/2.0.<br>
Via: SIP/2.0/UDP 75.74.203.73:60340;branch=z9hG4bK-d8754z-f6a3eadc786e7359-1---d8754z-;rport.<br>Max-Forwards: 70.<br>Contact: <sip:102@75.74.203.73:60340;transport=UDP>.<br>To: <<a href="mailto:sip%3A101@204.16.1.50" target="_blank">sip:101@204.16.1.50</a>;transport=UDP>.<br>
From: "102"<<a href="mailto:sip%3A102@204.16.1.50" target="_blank">sip:102@204.16.1.50</a>;transport=UDP>;tag=489f8f45.<br>Call-ID: ZGNhYTQzNjIyOGFkYWNhOWQ3ZmQ2ZDVkYjhiNGI4MGE..<br>CSeq: 1 INVITE.<br>Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE.<br>
Content-Type: application/sdp.<br>Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri.<br>User-Agent: Zoiper Communicator 2.04.10164 rev.10204.<br>Allow-Events: presence, kpml.<br>Content-Length: 352.<br>
<br>
<br></div>you can see the ip on signalling coming from 204.16.0.26 port 60340<br>on via you have <a href="http://75.74.203.73:60340" target="_blank">75.74.203.73:60340</a>, so you have a different ip address from signalling or via , in this case you will set the NAT variable, but check the invite below.<br>
<br>#<br>U <a href="http://204.16.0.26:5062" target="_blank">204.16.0.26:5062</a> -> <a href="http://204.16.1.50:5060" target="_blank">204.16.1.50:5060</a><br>INVITE <a href="mailto:sip%3A102@204.16.1.50" target="_blank">sip:102@204.16.1.50</a> SIP/2.0.<br>
Via: SIP/2.0/UDP 204.16.0.26:5062;branch=z9hG4bK1527256431.<br>
From: "Mike" <<a href="mailto:sip%3A101@204.16.1.50" target="_blank">sip:101@204.16.1.50</a>>;tag=1050377705.<br>To: <<a href="mailto:sip%3A102@204.16.1.50" target="_blank">sip:102@204.16.1.50</a>>.<br>
Call-ID: <a href="mailto:83821284@10.254.254.6" target="_blank">83821284@10.254.254.6</a>.<br>
CSeq: 1 INVITE.<br>Contact: <<a href="http://sip:101@204.16.0.26:5062" target="_blank">sip:101@204.16.0.26:5062</a>>.<br>Content-Type: application/sdp.<br>Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE.<br>
Max-Forwards: 70.<br>User-Agent: Yealink SIP-T20P 9.70.0.121.<br>Supported: replaces.<br>Allow-Events: talk,hold,conference,refer,check-sync.<br>Content-Length: 304<br><br></div>You have the same port on signalling and on VIA, in this case the rule will no match and variable will not be set and this is a phone behind a nat<br>
<br><br></div>so, you should try to remove the if where you call the rtpproxy offer and answer (just for test purpose)<br><br></div>you should increment you debug info too<br><br><div>/* uncomment the following lines to enable debugging */</div>
<div>#debug=6</div><div>#fork=no</div><div>#log_stderror=yes<br><br><br></div><div><br></div><br><div><div><div><div><div><div><div><div><div><br></div></div></div></div></div></div></div>
</div></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">2013/9/28 Rajesh Babu <span dir="ltr"><<a href="mailto:rajesh.babu@goodcoresoft.com" target="_blank">rajesh.babu@goodcoresoft.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div link="blue" vlink="purple" lang="EN-MY"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Hi,<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> I have attached the logs and my routing file @ <a href="http://pastebin.com/hu0bQGVw" target="_blank">http://pastebin.com/hu0bQGVw</a><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Please help me out in nailing this.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks <u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Rajesh<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm 0cm 0cm"><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US"> <a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a> [mailto:<a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a>] <b>On Behalf Of </b>Mike Tesliuk<br>
<b>Sent:</b> Friday, 27 September, 2013 11:25 PM</span></p><div><div><br><b>To:</b> OpenSIPS users mailling list<br><b>Subject:</b> Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork<u></u><u></u></div>
</div><p></p></div><div><div><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">If possible, paste your route file too<u></u><u></u></p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt">
<u></u> <u></u></p><div><p class="MsoNormal">2013/9/27 Mike Tesliuk <<a href="mailto:mike@ultra.net.br" target="_blank">mike@ultra.net.br</a>><u></u><u></u></p><div><div><p class="MsoNormal" style="margin-bottom:12.0pt">
start your opensips in debug mode, try to make the call, get all the message and paste in some pastebin website and show us the link<u></u><u></u></p></div></div><div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt">
<u></u> <u></u></p><div><p class="MsoNormal">2013/9/27 Rajesh Babu <<a href="mailto:rajesh.babu@goodcoresoft.com" target="_blank">rajesh.babu@goodcoresoft.com</a>><u></u><u></u></p><div><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I am getting Error 483, too many Hops, There is no other error messages i am getting. Please some one help me out in this</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><div><div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm 0cm 0cm">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US"> <a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a> [mailto:<a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a>] <b>On Behalf Of </b>Rajesh Babu<br>
<b>Sent:</b> Friday, 27 September, 2013 6:08 PM</span><u></u><u></u></p><div><p class="MsoNormal"><br><b>To:</b> 'OpenSIPS users mailling list'<br><b>Subject:</b> Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork<u></u><u></u></p>
</div></div></div><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">HI Mike,</span><u></u><u></u></p><div><p class="MsoNormal">
<span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> Now the RTP is up and i am getting this message on my logs</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">[root@centos64 rtpproxy-1.2.0]# tailf /var/log/messages</span><u></u><u></u></p><p class="MsoNormal">
<span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17800]: INFO:core:probe_max_sock_buff: using rcv buffer of 448 kb</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17810]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:<a href="http://10.10.10.123:7890" target="_blank">10.10.10.123:7890</a>> found, support for it enabled</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17812]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:<a href="http://10.10.10.123:7890" target="_blank">10.10.10.123:7890</a>> found, support for it enabled</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17815]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:<a href="http://10.10.10.123:7890" target="_blank">10.10.10.123:7890</a>> found, support for it enabled</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17808]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:<a href="http://10.10.10.123:7890" target="_blank">10.10.10.123:7890</a>> found, support for it enabled</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17811]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:<a href="http://10.10.10.123:7890" target="_blank">10.10.10.123:7890</a>> found, support for it enabled</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:<a href="http://10.10.10.123:7890" target="_blank">10.10.10.123:7890</a>> found, support for it enabled</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17814]: INFO:rtpproxy:rtpp_test: rtp proxy <udp:<a href="http://10.10.10.123:7890" target="_blank">10.10.10.123:7890</a>> found, support for it enabled</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]: WARNING:drouting:dr_load_routing_info: table "dr_rules" is empty</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 28 01:59:43 centos64 opensips: INFO:core:daemonize: pre-daemon process exiting with 0</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">But my test tool is not connecting back my server. Is there any mistake i am doing.</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks </span><u></u><u></u></p>
</div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Rajesh</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p>
<div><div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm 0cm 0cm"><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US"> <a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a> [<a href="mailto:users-bounces@lists.opensips.org" target="_blank">mailto:users-bounces@lists.opensips.org</a>] <b>On Behalf Of </b>Rajesh Babu<br>
<b>Sent:</b> Friday, 27 September, 2013 2:34 PM</span><u></u><u></u></p><div><p class="MsoNormal"><br><b>To:</b> 'OpenSIPS users mailling list'<br><b>Subject:</b> Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork<u></u><u></u></p>
</div></div></div><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Hi Mike,</span><u></u><u></u></p><div><p class="MsoNormal">
<span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> This is log i am geting wheni try to start the service</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">[root@centos64 rtpproxy-1.2.0]# tailf /var/log/messages</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12839]: WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has been disabled temporarily</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy Connection refused</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: ERROR:rtpproxy:send_rtpp_command: proxy <udp:localhost:7890> does not respond, disable it</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]: WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has been disabled temporarily</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy Connection refused</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: ERROR:rtpproxy:send_rtpp_command: proxy <udp:localhost:7890> does not respond, disable it</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]: WARNING:rtpproxy:rtpp_test: support for RTP proxy <udp:localhost:7890> has been disabled temporarily</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Sep 27 22:29:03 centos64 opensips: INFO:core:daemonize: pre-daemon process exiting with 0</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p></div><div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm 0cm 0cm">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US"> <a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a> [<a href="mailto:users-bounces@lists.opensips.org" target="_blank">mailto:users-bounces@lists.opensips.org</a>] <b>On Behalf Of </b>Mike Tesliuk<br>
<b>Sent:</b> Thursday, 26 September, 2013 10:25 PM</span><u></u><u></u></p><div><div><p class="MsoNormal"><br><b>To:</b> OpenSIPS users mailling list<br><b>Subject:</b> Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork<u></u><u></u></p>
</div></div></div><div><div><p class="MsoNormal"> <u></u><u></u></p><div><div><div><div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt">When you use the residential script almost all configuration come alredy working for this<u></u><u></u></p>
</div><p class="MsoNormal" style="margin-bottom:12.0pt">i have a tutorial (in portuguese ( i think that i should translate to english :) )) , where you can see a routing script working with nat<br><br><a href="http://opensips.com.br/wiki/index.php?title=Opensips_1.9" target="_blank">http://opensips.com.br/wiki/index.php?title=Opensips_1.9</a><u></u><u></u></p>
</div><p class="MsoNormal" style="margin-bottom:12.0pt">You can take a look at modules documentation too<br><br><a href="http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html" target="_blank">http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html</a><br>
<a href="http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html" target="_blank">http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html</a><u></u><u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt">
There is on this maillist too a lot of discussions about this, below you can see one case<br><br><a href="http://opensips.org/pipermail/users/2011-January/016130.html" target="_blank">http://opensips.org/pipermail/users/2011-January/016130.html</a><u></u><u></u></p>
</div><p class="MsoNormal" style="margin-bottom:12.0pt">If you get some information from an old version of opensips probably will be necessary to take a look on the module documentation to check about little diferences , but i think that this is the start point :)<u></u><u></u></p>
</div><p class="MsoNormal" style="margin-bottom:12.0pt">and if you is new to opensips i recommend to you the book about opensips ( <a href="http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book" target="_blank">http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book</a> )<u></u><u></u></p>
<div><div><div><div><div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt"> <u></u><u></u></p></div></div></div></div></div></div></div></div><div><p class="MsoNormal" style="margin-bottom:12.0pt"> <u></u><u></u></p>
<div><p class="MsoNormal">2013/9/26 Rajesh Babu <<a href="mailto:rajesh.babu@goodcoresoft.com" target="_blank">rajesh.babu@goodcoresoft.com</a>><u></u><u></u></p><div><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Hi Mike,</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> Thanks for the response, I am totally new to this world, can you please help me by directing to on how to configure links. It will be great. </span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thanks in advance</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Regards</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Rajesh</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p>
<div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm 0cm 0cm"><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US"> <a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a> [mailto:<a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a>] <b>On Behalf Of </b>Mike Tesliuk<br>
<b>Sent:</b> Thursday, 26 September, 2013 12:25 PM<br><b>To:</b> OpenSIPS users mailling list<br><b>Subject:</b> Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork</span><u></u><u></u></p>
</div><div><div><p class="MsoNormal"> <u></u><u></u></p><div><p class="MsoNormal">you should configure the nathelper and rtpproxy, this should help in you issue.<u></u><u></u></p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt">
<u></u><u></u></p><div><p class="MsoNormal">2013/9/26 Rajesh Babu <<a href="mailto:rajesh.babu@goodcoresoft.com" target="_blank">rajesh.babu@goodcoresoft.com</a>><u></u><u></u></p><div><div><div><p class="MsoNormal">
Hi,<u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal"> I am new to the OpenSIP world. I have installed a OpenSIP on my network. If i make a Call inside the network between two users i don’t have any issue, where as from outside the network, even though i can see the user registered in my server i am not able to call registered user (I see the user in my UL show listing). The call is established but i am not able to talk (Mean the audio and video are not getting transffered).<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">Where as messages are going fine without any issue. I guess it is because message transmit over XMPP where calls on SIP right.<u></u><u></u></p><p class="MsoNormal">
<u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p></div><p class="MsoNormal">I am really struck and i don’t know how to proceed, please help me out<u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">
<span style="color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="color:#1f497d">Thanks</span><u></u><u></u></p><p class="MsoNormal">
<span style="color:#1f497d">Rajesh</span><u></u><u></u></p></div></div><p class="MsoNormal" style="margin-bottom:12.0pt"><br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a><br>
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</div></div><p class="MsoNormal" style="margin-bottom:12.0pt"><br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a><br>
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</div></div><p class="MsoNormal" style="margin-bottom:12.0pt"><br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a><br>
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