Hello Alex, Yes, should have reinvites. I am getting randomly 404. Is this normal behaviour to specify outbound proxy on client on local network ( sounds bad ). Other wise is no rtp.
U 2014/03/26 11:59:36.207773 10.237.236.207:5060 -> 192.168.100.145:5062 INVITE sip:1200@192.168.100.145:5062 SIP/2.0. Record-Route: <sip:10.237.236.207;lr=on>. Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bK5a7e.03cc1cb9d02595ff6c7096253b4e6034.2. Via: SIP/2.0/UDP 10.237.236.212:63802;branch=z9hG4bK-d8754z-27d8d75f55eb3466-1---d8754z-;rport=63802. Max-Forwards: 16. Contact: <sip:1200@10.237.236.212:63802;transport=UDP>. To: <sip:1...@networklab.loc;transport=UDP>. From: <sip:1...@networklab.loc;transport=UDP>;tag=e145b359. Call-ID: ZTRiNGMzOWYxOTAyMjkyMGFjNjI0NjQzZGZmZDE4N2E.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri. User-Agent: Z 3.2.21357 r21367. Allow-Events: presence, kpml. Content-Length: 165. . v=0. o=Z 0 0 IN IP4 10.237.236.212. s=Z. c=IN IP4 10.237.236.212. t=0 0. m=audio 8000 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/03/26 11:59:36.211476 10.237.236.207:5062 -> 10.237.236.207:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bK5a7e.03cc1cb9d02595ff6c7096253b4e6034.2;received=10.237.236.207. Via: SIP/2.0/UDP 10.237.236.212:63802;branch=z9hG4bK-d8754z-27d8d75f55eb3466-1---d8754z-;rport=63802. From: <sip:1...@networklab.loc;transport=UDP>;tag=e145b359. To: <sip:1...@networklab.loc;transport=UDP>;tag=as75383b1b. Call-ID: ZTRiNGMzOWYxOTAyMjkyMGFjNjI0NjQzZGZmZDE4N2E.. CSeq: 1 INVITE. Server: Asterisk PBX 12.0.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. . ----- Original Message ----- From: "Alex Balashov" <abalas...@evaristesys.com> To: sr-users@lists.sip-router.org Sent: Wednesday, March 26, 2014 11:55:18 AM Subject: Re: [SR-Users] kamailio db On 03/26/2014 11:53 AM, Slava Bendersky wrote: > Hello Alex, > I added this section, right now I see mysql get updates. But still some > issue that is no rtp stream established. > When I place call between extensions I get dial tone and rings on > answer it dead. Well, that's progress! Kamailio is not involved in RTP, however[1]. Could it be that there is a network or transport-layer reachability issue between your endpoints? Or a firewall getting in the way, perhaps? -- Alex [1] It can control third-party, outboard RTP relays such as 'rtpproxy', though. But those are separate processes and pieces of software. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
