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:1200@networklab.loc;transport=UDP. From: sip:1200@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:1200@networklab.loc;transport=UDP;tag=e145b359. To: sip:1200@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" abalashov@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.