Sorry Bogdan but now my setup become a bit differente, i have the same servers , Opensips+Asterisk in EC2 amazon (same LAN) and Cisco gateway outside conected through public_ip to Opensips.<br>The SIP signalling works well but i have just oneway audio cause asterisk send private ip on the reply to opensips invite (in same LAN) and opensips forward that private ip to Cisco. So asterisk know the public ip of cisco to establish rtp traffic but cisco don´t. ¿how can i solve this problem ? ¿there is anyway to change the rtp ip in the invite's reply ?<br>
Best Regards!!<br><br><br>opensips.cfg:<br>route{<br><br> if (!mf_process_maxfwd_header("10")) {<br> sl_send_reply("483","looping");<br> exit;<br> }<br>
if ($rU==NULL) {<br> sl_send_reply("484","Address Incomplete");<br> exit;<br> }<br> if (!has_totag()) {<br> record_route_preset(" Opensips public ip ");<br>
xlog("route recorded \n");<br> } else {<br> loose_route();<br> t_relay();<br> exit;<br> }<br> if ( is_method("CANCEL") ) {<br>
if ( t_check_trans() )<br> t_relay();<br> exit;<br> }<br> if (!is_method("INVITE")) {<br> send_reply("405","Method Not Allowed");<br>
exit;<br> }<br> if (method=="INVITE") {<br> load_balance("1","calls");<br> }<br><br> if ($retcode<0) {<br> sl_send_reply("500","Service full");<br>
exit;<br> }<br><br> xlog("Selected destination is: $du\n");<br><br> if (!t_relay()) {<br> sl_reply_error();<br> }<br>}<br> <br>######################################################################################################<br>
<br>U 2010/12/03 13:00:27.034603 <a href="http://80.65.13.238:65071">80.65.13.238:65071</a> -> <a href="http://10.229.123.198:5060">10.229.123.198:5060</a><br>INVITE <a href="http://sip:911126667@x.911126667.opensips.lab.egtelecom.es:5060">sip:911126667@x.911126667.opensips.lab.egtelecom.es:5060</a> SIP/2.0.<br>
Date: Fri, 03 Dec 2010 12:04:32 GMT.<br>Call-Info: <sip:<a href="http://80.65.13.238:5060">80.65.13.238:5060</a>>;method="NOTIFY;Event=telephone-event;Duration=2000".<br>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER.<br>
From: <<a href="mailto:sip%3A911873699@80.65.13.238">sip:911873699@80.65.13.238</a>>;tag=274FBBA0-208D.<br>Allow-Events: telephone-event.<br>Supported: 100rel,timer,resource-priority,replaces,sdp-anat.<br>Min-SE: 1800.<br>
Remote-Party-ID: <<a href="mailto:sip%3A911873699@80.65.13.238">sip:911873699@80.65.13.238</a>>;party=calling;screen=yes;privacy=off.<br>Cisco-Guid: 1378169425-4262203871-3197108258-2438471722.<br>Timestamp: 1291377872.<br>
Content-Length: 269.<br>User-Agent: Cisco-SIPGateway/IOS-12.x.<br>To: <<a href="mailto:sip%3A911126667@x.911126667.opensips.lab.egtelecom.es">sip:911126667@x.911126667.opensips.lab.egtelecom.es</a>>.<br>Contact: <<a href="http://sip:911873699@80.65.13.238:5060">sip:911873699@80.65.13.238:5060</a>>.<br>
Expires: 180.<br>Content-Disposition: session;handling=required.<br>Content-Type: application/sdp.<br>Call-ID: <a href="mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238">5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238</a>.<br>
Via: SIP/2.0/UDP 80.65.13.238:5060;x-route-tag="<a href="mailto:cid%3AOrange@80.65.13.238">cid:Orange@80.65.13.238</a>";branch=z9hG4bK1EDAC71C9A.<br>CSeq: 101 INVITE.<br>Max-Forwards: 70.<br>.<br>v=0.<br>o=CiscoSystemsSIP-GW-UserAgent 849 9795 IN IP4 80.65.13.238.<br>
s=SIP Call.<br>c=IN IP4 80.65.13.238.<br>t=0 0.<br>m=audio 23660 RTP/AVP 18 101.<br>c=IN IP4 80.65.13.238.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br>
<br><br>U 2010/12/03 13:00:27.035190 <a href="http://10.229.123.198:5060">10.229.123.198:5060</a> -> <a href="http://80.65.13.238:5060">80.65.13.238:5060</a><br>SIP/2.0 100 Giving a try.<br>From: <<a href="mailto:sip%3A911873699@80.65.13.238">sip:911873699@80.65.13.238</a>>;tag=274FBBA0-208D.<br>
To: <<a href="mailto:sip%3A911126667@x.911126667.opensips.lab.egtelecom.es">sip:911126667@x.911126667.opensips.lab.egtelecom.es</a>>.<br>Call-ID: <a href="mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238">5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238</a>.<br>
Via: SIP/2.0/UDP 80.65.13.238:5060;x-route-tag="<a href="mailto:cid%3AOrange@80.65.13.238">cid:Orange@80.65.13.238</a>";branch=z9hG4bK1EDAC71C9A.<br>CSeq: 101 INVITE.<br>Server: OpenSIPS (1.6.3-notls (i386/linux)).<br>
Content-Length: 0.<br>.<br><br><br>U 2010/12/03 13:00:27.035263 <a href="http://10.229.123.198:5060">10.229.123.198:5060</a> -> <a href="http://10.228.26.150:5060">10.228.26.150:5060</a><br>INVITE <a href="http://sip:911126667@x.911126667.opensips.lab.egtelecom.es:5060">sip:911126667@x.911126667.opensips.lab.egtelecom.es:5060</a> SIP/2.0.<br>
Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.<br>Date: Fri, 03 Dec 2010 12:04:32 GMT.<br>Call-Info: <sip:<a href="http://80.65.13.238:5060">80.65.13.238:5060</a>>;method="NOTIFY;Event=telephone-event;Duration=2000".<br>
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER.<br>From: <<a href="mailto:sip%3A911873699@80.65.13.238">sip:911873699@80.65.13.238</a>>;tag=274FBBA0-208D.<br>Allow-Events: telephone-event.<br>
Supported: 100rel,timer,resource-priority,replaces,sdp-anat.<br>Min-SE: 1800.<br>Remote-Party-ID: <<a href="mailto:sip%3A911873699@80.65.13.238">sip:911873699@80.65.13.238</a>>;party=calling;screen=yes;privacy=off.<br>
Cisco-Guid: 1378169425-4262203871-3197108258-2438471722.<br>Timestamp: 1291377872.<br>Content-Length: 269.<br>User-Agent: Cisco-SIPGateway/IOS-12.x.<br>To: <<a href="mailto:sip%3A911126667@x.911126667.opensips.lab.egtelecom.es">sip:911126667@x.911126667.opensips.lab.egtelecom.es</a>>.<br>
Contact: <<a href="http://sip:911873699@80.65.13.238:5060">sip:911873699@80.65.13.238:5060</a>>.<br>Expires: 180.<br>Content-Disposition: session;handling=required.<br>Content-Type: application/sdp.<br>Call-ID: <a href="mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238">5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238</a>.<br>
Via: SIP/2.0/UDP 46.51.135.212;branch=z9hG4bK1a6f.13422624.0.<br>Via: SIP/2.0/UDP 80.65.13.238:5060;x-route-tag="<a href="mailto:cid%3AOrange@80.65.13.238">cid:Orange@80.65.13.238</a>";branch=z9hG4bK1EDAC71C9A.<br>
CSeq: 101 INVITE.<br>Max-Forwards: 69.<br>.<br>v=0.<br>o=CiscoSystemsSIP-GW-UserAgent 849 9795 IN IP4 80.65.13.238.<br>s=SIP Call.<br>c=IN IP4 80.65.13.238.<br>t=0 0.<br>m=audio 23660 RTP/AVP 18 101.<br>c=IN IP4 80.65.13.238.<br>
a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br><br><br>U 2010/12/03 13:00:27.036250 <a href="http://10.228.26.150:5060">10.228.26.150:5060</a> -> <a href="http://10.229.123.198:5060">10.229.123.198:5060</a><br>
SIP/2.0 100 Trying.<br>Via: SIP/2.0/UDP 46.51.135.212;branch=z9hG4bK1a6f.13422624.0;received=10.229.123.198.<br>Via: SIP/2.0/UDP 80.65.13.238:5060;x-route-tag="<a href="mailto:cid%3AOrange@80.65.13.238">cid:Orange@80.65.13.238</a>";branch=z9hG4bK1EDAC71C9A.<br>
Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.<br>From: <<a href="mailto:sip%3A911873699@80.65.13.238">sip:911873699@80.65.13.238</a>>;tag=274FBBA0-208D.<br>To: <<a href="mailto:sip%3A911126667@x.911126667.opensips.lab.egtelecom.es">sip:911126667@x.911126667.opensips.lab.egtelecom.es</a>>.<br>
Call-ID: <a href="mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238">5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238</a>.<br>CSeq: 101 INVITE.<br>Server: Asterisk PBX 1.6.2.13.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.<br>
Supported: replaces, timer.<br>Require: timer.<br>Session-Expires: 1800;refresher=uas.<br>Contact: <<a href="mailto:sip%3A911126667@10.228.26.150">sip:911126667@10.228.26.150</a>>.<br>Content-Length: 0.<br>.<br><br>
<br>U 2010/12/03 13:00:27.235884 <a href="http://10.228.26.150:5060">10.228.26.150:5060</a> -> <a href="http://10.229.123.198:5060">10.229.123.198:5060</a><br>SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 46.51.135.212;branch=z9hG4bK1a6f.13422624.0;received=10.229.123.198.<br>
Via: SIP/2.0/UDP 80.65.13.238:5060;x-route-tag="<a href="mailto:cid%3AOrange@80.65.13.238">cid:Orange@80.65.13.238</a>";branch=z9hG4bK1EDAC71C9A.<br>Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.<br>
From: <<a href="mailto:sip%3A911873699@80.65.13.238">sip:911873699@80.65.13.238</a>>;tag=274FBBA0-208D.<br>To: <<a href="mailto:sip%3A911126667@x.911126667.opensips.lab.egtelecom.es">sip:911126667@x.911126667.opensips.lab.egtelecom.es</a>>;tag=as33981ab2.<br>
Call-ID: <a href="mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238">5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238</a>.<br>CSeq: 101 INVITE.<br>Server: Asterisk PBX 1.6.2.13.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.<br>
Supported: replaces, timer.<br>Require: timer.<br>Session-Expires: 1800;refresher=uas.<br>Contact: <<a href="mailto:sip%3A911126667@10.228.26.150">sip:911126667@10.228.26.150</a>>.<br>Content-Type: application/sdp.<br>
Content-Length: 262.<br>.<br>v=0.<br>o=root 1270939673 1270939673 IN IP4 10.228.26.150.<br>s=Asterisk PBX 1.6.2.13.<br>c=IN IP4 10.228.26.150.<br>t=0 0.<br>m=audio 10532 RTP/AVP 18 101.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>
a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br>a=sendrecv.<br><br><br>U 2010/12/03 13:00:27.236908 <a href="http://10.229.123.198:5060">10.229.123.198:5060</a> -> <a href="http://80.65.13.238:5060">80.65.13.238:5060</a><br>
SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 80.65.13.238:5060;x-route-tag="<a href="mailto:cid%3AOrange@80.65.13.238">cid:Orange@80.65.13.238</a>";branch=z9hG4bK1EDAC71C9A.<br>Record-Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.<br>
From: <<a href="mailto:sip%3A911873699@80.65.13.238">sip:911873699@80.65.13.238</a>>;tag=274FBBA0-208D.<br>To: <<a href="mailto:sip%3A911126667@x.911126667.opensips.lab.egtelecom.es">sip:911126667@x.911126667.opensips.lab.egtelecom.es</a>>;tag=as33981ab2.<br>
Call-ID: <a href="mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238">5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238</a>.<br>CSeq: 101 INVITE.<br>Server: Asterisk PBX 1.6.2.13.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.<br>
Supported: replaces, timer.<br>Require: timer.<br>Session-Expires: 1800;refresher=uas.<br>Contact: <<a href="mailto:sip%3A911126667@10.228.26.150">sip:911126667@10.228.26.150</a>>.<br>Content-Type: application/sdp.<br>
Content-Length: 262.<br>.<br>v=0.<br>o=root 1270939673 1270939673 IN IP4 10.228.26.150.<br>s=Asterisk PBX 1.6.2.13.<br>c=IN IP4 10.228.26.150.<br>t=0 0.<br>m=audio 10532 RTP/AVP 18 101.<br>a=rtpmap:18 G729/8000.<br>a=fmtp:18 annexb=no.<br>
a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-16.<br>a=ptime:20.<br>a=sendrecv.<br><br><br>U 2010/12/03 13:00:27.294728 <a href="http://80.65.13.238:65071">80.65.13.238:65071</a> -> <a href="http://10.229.123.198:5060">10.229.123.198:5060</a><br>
ACK <a href="http://sip:911126667@10.228.26.150:5060">sip:911126667@10.228.26.150:5060</a> SIP/2.0.<br>Route: <sip:46.51.135.212;lr=on;did=fc7.6548ee66>.<br>Date: Fri, 03 Dec 2010 12:04:32 GMT.<br>From: <<a href="mailto:sip%3A911873699@80.65.13.238">sip:911873699@80.65.13.238</a>>;tag=274FBBA0-208D.<br>
Allow-Events: telephone-event.<br>Content-Length: 0.<br>To: <<a href="mailto:sip%3A911126667@x.911126667.opensips.lab.egtelecom.es">sip:911126667@x.911126667.opensips.lab.egtelecom.es</a>>;tag=as33981ab2.<br>Call-ID: <a href="mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238">5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238</a>.<br>
Via: SIP/2.0/UDP 80.65.13.238:5060;x-route-tag="<a href="mailto:cid%3AOrange@80.65.13.238">cid:Orange@80.65.13.238</a>";branch=z9hG4bK1EDAC8B3D.<br>CSeq: 101 ACK.<br>Max-Forwards: 70.<br>.<br><br><br>U 2010/12/03 13:00:27.295705 <a href="http://10.229.123.198:5060">10.229.123.198:5060</a> -> <a href="http://10.228.26.150:5060">10.228.26.150:5060</a><br>
ACK <a href="http://sip:911126667@10.228.26.150:5060">sip:911126667@10.228.26.150:5060</a> SIP/2.0.<br>Date: Fri, 03 Dec 2010 12:04:32 GMT.<br>From: <<a href="mailto:sip%3A911873699@80.65.13.238">sip:911873699@80.65.13.238</a>>;tag=274FBBA0-208D.<br>
Allow-Events: telephone-event.<br>Content-Length: 0.<br>To: <<a href="mailto:sip%3A911126667@x.911126667.opensips.lab.egtelecom.es">sip:911126667@x.911126667.opensips.lab.egtelecom.es</a>>;tag=as33981ab2.<br>Call-ID: <a href="mailto:5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238">5225CE79-FE0C11DF-85C6D89D-59E864DE@80.65.13.238</a>.<br>
Via: SIP/2.0/UDP 46.51.135.212;branch=z9hG4bK1a6f.13422624.2.<br>Via: SIP/2.0/UDP 80.65.13.238:5060;x-route-tag="<a href="mailto:cid%3AOrange@80.65.13.238">cid:Orange@80.65.13.238</a>";branch=z9hG4bK1EDAC8B3D.<br>
CSeq: 101 ACK.<br>Max-Forwards: 69.<br><br><div class="gmail_quote"> 2010/12/2 Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a>></span><br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi Nawfel,<br>
<br>
The problem is in one of the end points as for a 200 OK calls, the 200 reply and the ACK is end-2-end.<br>
<br>
If you have a trace, maybe I can help you to see if there is a signalling problem.<br>
<br>
Regards,<br>
Bogdan<div class="im"><br>
<br>
Nawfel Oujdi wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hello!!<br>
I m new in opensips and i m testing the load balancer cause i need it to balance calls between 4 asterisk.For the start i make the following scenario <br>
Cisco gateway inbound ------> opensips ------> asterisk ---------> Cisco gateway outbound<br>
when the call comes to the opensips, the load_balancer forward the call correctly to my asterisk but the call hangs up after 15 seg approximately.When i did a ngrep for the sip traffic in opensips, i realized that cisco gateway inbound never sent the ACK for 200 OK to opensips .<br>
In the Cisco's logs i saw that the reply of 200 ok is sent directly to public ip of asterisk but never to opensips server so asterisk still waiting for the ACK from opensips.<br>
In the same way opensips never receive the BYE packet and the load never decrease when the call is hanging up.<br>
<br>
Cisco gateway opensips asterisk<br>
---invite---> <--trying---- ---invite---> <---trying---<br>
<----200OK---<br>
<---200 OK--- <----200OK---<br>
<---200 OK--- <----200OK---<br>
<---200 OK--- <----200OK---<br>
<---200 OK--- Please can somebady help me to understand what cause that?<br>
<br>
Best Regards!! <br>
</blockquote>
<br>
<br></div>
-- <br>
Bogdan-Andrei Iancu<br>
OpenSIPS Bootcamp<br>
15 - 19 November 2010, Edison, New Jersey, USA<br>
<a href="http://www.voice-system.ro" target="_blank">www.voice-system.ro</a><br>
<br>
<br>
_______________________________________________<br>
Users mailing list<br>
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<a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
</blockquote></div><br><br clear="all"><br>-- <br><span style="font-size: 13px; font-family: arial,sans-serif; border-collapse: collapse;">
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<div style="font-weight: bold; font-size: 15px; color: rgb(15, 120, 180); font-family: Arial;">Nawfel Oujdi</div>
<div style="font-size: 12px; color: rgb(100, 101, 103); font-family: Arial;"><b>Ingeniero VoIP</b></div>
<div style="font-size: 12px; color: rgb(100, 101, 103); font-family: Arial;"><a style="color: rgb(15, 120, 180);" href="mailto:noujdi@egtelecom.es" target="_blank">noujdi@egtelecom.es</a></div></div>
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