<html><body><div style="font-family: lucida console,sans-serif; font-size: 12pt; color: #000000"><div>Hello Alex,<br></div><div>Yes, should have reinvites. I am getting randomly 404.</div><div>Is this normal behaviour to specify outbound proxy on client on local network ( sounds bad ). <br></div><div>Other wise is no rtp.<br></div><div><br></div><div><br></div><div>U 2014/03/26 11:59:36.207773 10.237.236.207:5060 -> 192.168.100.145:5062<br>INVITE sip:1200@192.168.100.145:5062 SIP/2.0.<br>Record-Route: <sip:10.237.236.207;lr=on>.<br>Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bK5a7e.03cc1cb9d02595ff6c7096253b4e6034.2.<br>Via: SIP/2.0/UDP 10.237.236.212:63802;branch=z9hG4bK-d8754z-27d8d75f55eb3466-1---d8754z-;rport=63802.<br>Max-Forwards: 16.<br>Contact: <sip:1200@10.237.236.212:63802;transport=UDP>.<br>To: <sip:1200@networklab.loc;transport=UDP>.<br>From: <sip:1200@networklab.loc;transport=UDP>;tag=e145b359.<br>Call-ID: ZTRiNGMzOWYxOTAyMjkyMGFjNjI0NjQzZGZmZDE4N2E..<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, timer, X-cisco-serviceuri.<br>User-Agent: Z 3.2.21357 r21367.<br>Allow-Events: presence, kpml.<br>Content-Length: 165.<br>.<br>v=0.<br>o=Z 0 0 IN IP4 10.237.236.212.<br>s=Z.<br>c=IN IP4 10.237.236.212.<br>t=0 0.<br>m=audio 8000 RTP/AVP 0 101.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-15.<br>a=sendrecv.<br><br><br>U 2014/03/26 11:59:36.211476 10.237.236.207:5062 -> 10.237.236.207:5060<br>SIP/2.0 404 Not Found.<br>Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bK5a7e.03cc1cb9d02595ff6c7096253b4e6034.2;received=10.237.236.207.<br>Via: SIP/2.0/UDP 10.237.236.212:63802;branch=z9hG4bK-d8754z-27d8d75f55eb3466-1---d8754z-;rport=63802.<br>From: <sip:1200@networklab.loc;transport=UDP>;tag=e145b359.<br>To: <sip:1200@networklab.loc;transport=UDP>;tag=as75383b1b.<br>Call-ID: ZTRiNGMzOWYxOTAyMjkyMGFjNjI0NjQzZGZmZDE4N2E..<br>CSeq: 1 INVITE.<br>Server: Asterisk PBX 12.0.0.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.<br>Supported: replaces, timer.<br>Content-Length: 0.<br>.<br><br></div><div><br></div><div> <br></div><div><br></div><hr id="zwchr"><div style="color:#000;font-weight:normal;font-style:normal;text-decoration:none;font-family:Helvetica,Arial,sans-serif;font-size:12pt;"><b>From: </b>"Alex Balashov" <abalashov@evaristesys.com><br><b>To: </b>sr-users@lists.sip-router.org<br><b>Sent: </b>Wednesday, March 26, 2014 11:55:18 AM<br><b>Subject: </b>Re: [SR-Users] kamailio db<br><div><br></div>On 03/26/2014 11:53 AM, Slava Bendersky wrote:<br>> Hello Alex,<br>> I added this section, right now I see mysql get updates. But still some<br>> issue that is no rtp stream established.<br>> When I place call between extensions I get dial tone and rings on<br>> answer it dead.<br><div><br></div>Well, that's progress!<br><div><br></div>Kamailio is not involved in RTP, however[1].<br><div><br></div>Could it be that there is a network or transport-layer reachability <br>issue between your endpoints? Or a firewall getting in the way, perhaps?<br><div><br></div>-- Alex<br><div><br></div>[1] It can control third-party, outboard RTP relays such as 'rtpproxy', <br>though. But those are separate processes and pieces of software.<br><div><br></div>-- <br>Alex Balashov - Principal<br>Evariste Systems LLC<br>235 E Ponce de Leon Ave<br>Suite 106<br>Decatur, GA 30030<br>United States<br>Tel: +1-678-954-0670<br>Web: http://www.evaristesys.com/, http://www.alexbalashov.com/<br><div><br></div>_______________________________________________<br>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>sr-users@lists.sip-router.org<br>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br></div><div><br></div></div></body></html>