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.
.

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.
--
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/
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