[SR-Users] kamailio db

[Slava Bendersky]

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 at 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 at 10.237.236.212:63802;transport=UDP>. 
To: <sip:1200 at networklab.loc;transport=UDP>. 
From: <sip:1200 at 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 at networklab.loc;transport=UDP>;tag=e145b359. 
To: <sip:1200 at 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 at evaristesys.com> 
To: sr-users at 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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