Hi,<div><br><div>You don't give the logs and sip traces - its no fun for you or anyone here. I recommend you to change your FreeSwitch/Asterisk settings to stop re-invites (canreinvite=no) and in newer version (directmedia=no) - See if that helps.</div>
<div> </div><div>I'll again repeat that either you don't engage RTPproxy in the call at all, or you send the Media via the RTPproxy by setting your A-leg & B-leg telephony engines. that'll solve your problem</div>
<div><br></div><div>I hope you can handle it from here now.</div><div><br></div><div>Regards,</div><div>Sammy</div><div><br><br><div class="gmail_quote">On Wed, Jul 4, 2012 at 10:15 PM, Rodrigo Ferreira <span dir="ltr"><<a href="mailto:rodrigo.ferreira@vipway.net.br" target="_blank">rodrigo.ferreira@vipway.net.br</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div dir="ltr">
<div style="font-size:12pt;font-family:'Calibri'">
<div>I’m trying to debug those calls using wireshark and VoIP calls analisys ...
</div>
<div> </div>
<div>And i guess it might be a network problem, maybe some packets being dropped
.. Is there any chance of being that? </div>
<div>Because my Opensips box and my FreePBX (asterisk) they are in the same
network, just passing through a switch, among with other servers (including
another opensips who is giving me the same problem)</div>
<div style="font-size:small;font-style:normal;text-decoration:none;font-family:'Calibri';display:inline;font-weight:normal">
<div style="FONT:10pt tahoma">
<div><font size="3" face="Calibri"></font> </div>
<div style="BACKGROUND:#f5f5f5">
<div><b>From:</b> <a title="rodrigo.ferreira@vipway.net.br" href="mailto:rodrigo.ferreira@vipway.net.br" target="_blank">Rodrigo Ferreira</a> </div>
<div><b>Sent:</b> Wednesday, July 04, 2012 9:10 AM</div><div><div class="h5">
<div><b>To:</b> <a title="users@lists.opensips.org" href="mailto:users@lists.opensips.org" target="_blank">OpenSIPS users mailling list</a> </div>
<div><b>Subject:</b> Re: [OpenSIPS-Users] Calls being
disconnected</div></div></div></div></div>
<div> </div></div>
<div style="font-size:small;font-style:normal;text-decoration:none;font-family:'Calibri';display:inline;font-weight:normal"><div><div class="h5">
<div dir="ltr">
<div style="font-size:12pt;font-family:'Calibri'">
<div>That was my bad, its different IPs, but I wrote it wrong.</div>
<div> </div>
<div>I will get some new traces today, and I will write the IPs correctly </div>
<div style="font-size:small;font-style:normal;text-decoration:none;font-family:'Calibri';display:inline;font-weight:normal">
<div style="FONT:10pt tahoma">
<div> </div>
<div style="BACKGROUND:#f5f5f5">
<div><b>From:</b> <a title="govoiper@gmail.com" href="mailto:govoiper@gmail.com" target="_blank">SamyGo</a> </div>
<div><b>Sent:</b> Wednesday, July 04, 2012 1:54 AM</div>
<div><b>To:</b> <a title="users@lists.opensips.org" href="mailto:users@lists.opensips.org" target="_blank">OpenSIPS users mailling list</a> </div>
<div><b>Subject:</b> Re: [OpenSIPS-Users] Calls being
disconnected</div></div></div>
<div> </div></div>
<div style="font-size:small;font-style:normal;text-decoration:none;font-family:'Calibri';display:inline;font-weight:normal">I
will definitely try to read the SIP traces. I love trying to solve the puzzle
but the very first line !!
<div> </div>
<div><font face="courier new, monospace">U 201.71.XXX.XXX:5060 ->
201.71.XXX.XXX:5060</font></div>
<div><font face="courier new, monospace"><br></font></div>
<div><font face="arial, helvetica, sans-serif">Is it the same server originating
SIP INVITE sending to the same server !! if not please differentiate the two
addresses like.</font></div>
<div><font face="arial, helvetica, sans-serif"><br></font><span style="FONT-FAMILY:'courier new',monospace">U 201.71.XXX.XXX:5060 ->
201.71.XXX.YYY:5060</span> </div>
<div> </div>
<div>Goodness , its FBX,OpenSIPS, gatewaySTS, and even Asterisk 1.8 -- are they
all using the same IP !!</div>
<div> </div>
<div>Caller(FPBX)<--->OpenSIPS+RTPproxy<---->gatewaySTS<---->Asterisk</div>
<div>
||<============================RTP===============>||</div>
<div> </div>
<div>Please carefully use different IP addresses of all the SIP servers/UACs in
your particular call scenario. What I'm still suspecting with more intensity is
that somehow the RTPproxy is called but Asterisk and FPBX negotiate media
directly in SDPs and bam..RTPproxy gets angry at this insulting attitude and
disconnects the call !! That's my guess.</div>
<div> </div>
<div>OpenSIPS logs along with the new SIP traces will help clarify the real
cause.</div>
<div> </div>
<div>Thanks</div>
<div>Sammy</div>
<div>
<div> </div>
<div class="gmail_quote">On Tue, Jul 3, 2012 at 7:23 PM, Rodrigo Ferreira <span dir="ltr"><<a href="mailto:rodrigo.ferreira@vipway.net.br" target="_blank">rodrigo.ferreira@vipway.net.br</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote">
<div dir="ltr">
<div dir="ltr">
<div style="FONT-FAMILY:'Calibri';FONT-SIZE:12pt">
<div>I’m attaching the log that I got from a call that was disconnected</div>
<div style="FONT-STYLE:normal;DISPLAY:inline;FONT-FAMILY:'Calibri';FONT-SIZE:small;FONT-WEIGHT:normal;TEXT-DECORATION:none">
<div style="FONT:10pt tahoma">
<div> </div>
<div style="BACKGROUND:#f5f5f5">
<div><b>From:</b> <a title="rodrigo.ferreira@vipway.net.br" href="mailto:rodrigo.ferreira@vipway.net.br" target="_blank">Rodrigo
Ferreira</a> </div>
<div><b>Sent:</b> Tuesday, July 03, 2012 11:09 AM</div>
<div>
<div>
<div><b>To:</b> <a title="users@lists.opensips.org" href="mailto:users@lists.opensips.org" target="_blank">OpenSIPS users mailling
list</a> </div>
<div><b>Subject:</b> Re: [OpenSIPS-Users] Calls being
disconnected</div></div></div></div></div>
<div> </div></div>
<div>
<div>
<div style="FONT-STYLE:normal;DISPLAY:inline;FONT-FAMILY:'Calibri';FONT-SIZE:small;FONT-WEIGHT:normal;TEXT-DECORATION:none">
<div dir="ltr">
<div style="FONT-FAMILY:'Calibri';FONT-SIZE:12pt">
<div>I’m getting the logs, but let me ask you something, force_rport will help
me? </div>
<div style="FONT-STYLE:normal;DISPLAY:inline;FONT-FAMILY:'Calibri';FONT-SIZE:small;FONT-WEIGHT:normal;TEXT-DECORATION:none">
<div style="FONT:10pt tahoma">
<div> </div>
<div style="BACKGROUND:#f5f5f5">
<div><b>From:</b> <a title="govoiper@gmail.com" href="mailto:govoiper@gmail.com" target="_blank">SamyGo</a> </div>
<div><b>Sent:</b> Tuesday, July 03, 2012 8:18 AM</div>
<div><b>To:</b> <a title="users@lists.opensips.org" href="mailto:users@lists.opensips.org" target="_blank">OpenSIPS users mailling
list</a> </div>
<div><b>Subject:</b> Re: [OpenSIPS-Users] Calls being
disconnected</div></div></div>
<div> </div></div>
<div style="FONT-STYLE:normal;DISPLAY:inline;FONT-FAMILY:'Calibri';FONT-SIZE:small;FONT-WEIGHT:normal;TEXT-DECORATION:none">Specifically
talking about one where call gets dropped - what is the network scenario
!? Make one such call and take a sip trace on the opensips server using
tcpdump or sipgrep. Share the pcap. As well as the opensips logs please.
<div>
<div> </div>
<div> </div>
<div>
<div class="gmail_quote">On Tue, Jul 3, 2012 at 4:12 PM, Rodrigo Ferreira <span dir="ltr"><<a href="mailto:rodrigo.ferreira@vipway.net.br" target="_blank">rodrigo.ferreira@vipway.net.br</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote">
<div dir="ltr">
<div dir="ltr">
<div style="FONT-FAMILY:'Calibri';FONT-SIZE:12pt">
<div>I’ve a lot of different scenarios through my network, I have clients
behind NAT, from other networks, because I work for a telephony
company.</div>
<div style="FONT-STYLE:normal;DISPLAY:inline;FONT-FAMILY:'Calibri';FONT-SIZE:small;FONT-WEIGHT:normal;TEXT-DECORATION:none">
<div style="FONT:10pt tahoma">
<div><font size="3" face="Calibri"></font> </div>
<div style="BACKGROUND:#f5f5f5">
<div><b>From:</b> <a title="govoiper@gmail.com" href="mailto:govoiper@gmail.com" target="_blank">SamyGo</a> </div>
<div><b>Sent:</b> Tuesday, July 03, 2012 8:02 AM</div>
<div>
<div><b>To:</b> <a title="users@lists.opensips.org" href="mailto:users@lists.opensips.org" target="_blank">OpenSIPS users mailling
list</a> </div></div>
<div>
<div>
<div><b>Subject:</b> Re: [OpenSIPS-Users] Calls being
disconnected</div></div></div></div></div>
<div> </div></div>
<div>
<div>
<div style="FONT-STYLE:normal;DISPLAY:inline;FONT-FAMILY:'Calibri';FONT-SIZE:small;FONT-WEIGHT:normal;TEXT-DECORATION:none">RTPproxy
- thats what I suspected. What i'm guessing is that RTPproxy is engaged but
somehow the end points receive each other's Direct IPs in SDP...hmm...tell
me if your both end points are on the same LAN and server is placed
somewhere in internet !!<br><br>
<div class="gmail_quote">On Tue, Jul 3, 2012 at 3:58 PM, Rodrigo Ferreira
<span dir="ltr"><<a href="mailto:rodrigo.ferreira@vipway.net.br" target="_blank">rodrigo.ferreira@vipway.net.br</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote">
<div dir="ltr">
<div dir="ltr">
<div style="FONT-FAMILY:'Calibri';FONT-SIZE:12pt">
<div>I will take a look at my rtp .. I’m using rtpproxy on my box, I will
try to get some logs today too</div>
<div style="FONT-STYLE:normal;DISPLAY:inline;FONT-FAMILY:'Calibri';FONT-SIZE:small;FONT-WEIGHT:normal;TEXT-DECORATION:none">
<div style="FONT:10pt tahoma">
<div> </div>
<div style="BACKGROUND:#f5f5f5">
<div><b>From:</b> <a title="govoiper@gmail.com" href="mailto:govoiper@gmail.com" target="_blank">SamyGo</a> </div>
<div><b>Sent:</b> Tuesday, July 03, 2012 4:34 AM</div>
<div><b>To:</b> <a title="aamir_ryu@yahoo.com" href="mailto:aamir_ryu@yahoo.com" target="_blank">aamir chougule</a> ; <a title="users@lists.opensips.org" href="mailto:users@lists.opensips.org" target="_blank">OpenSIPS users mailling list</a> </div>
<div><b>Subject:</b> Re: [OpenSIPS-Users] Calls being
disconnected</div></div></div>
<div> </div></div>
<div style="FONT-STYLE:normal;DISPLAY:inline;FONT-FAMILY:'Calibri';FONT-SIZE:small;FONT-WEIGHT:normal;TEXT-DECORATION:none">
<div>
<div>Besides logs and any sip traces as Aamir stated it'd be hard to pin
point anything. One of the big reason for automatically dropping an
established call specially after a specific amount of time i.e 30
seconds..thats because of lack of RTPs flowing through the server and if
media-relaying tools are used they will send hangup to both ends, even
though both ends have their RTP flowing <u>directly</u>.<br><br>Regards,
<div>Sammy</div>
<div>
<div> </div>
<div class="gmail_quote">On Tue, Jul 3, 2012 at 12:21 PM, aamir chougule
<span dir="ltr"><<a href="mailto:aamir_ryu@yahoo.com" target="_blank">aamir_ryu@yahoo.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote">
<div>
<div style="FONT-FAMILY:times new roman,new york,times,serif;FONT-SIZE:12pt">
<div><span>Hi Rodrigo,</span></div>
<div><br><span></span></div>
<div><span>The only which I know is after the call gets connected i.e.
200OK from the other end there should always be an ACK for that 200OK,
if there is no ACK for the 200OK then the call will get disconnected
after sometime.</span></div>
<div><br><span></span></div>
<div><span>Can you please give us the logs for the call by tracing it
through ngrep tool because without any sip traces or opensips log we
can't tell what is going on within the box.</span></div>
<div> </div>
<div style="FONT-FAMILY:times new roman,new york,times,serif;COLOR:rgb(3,61,33);FONT-WEIGHT:bold">Regards,<br><br>Aamir
Chougule<span></span></div>
<div style="BACKGROUND-COLOR:transparent;FONT-STYLE:normal;FONT-FAMILY:arial,helvetica,clean,sans-serif;FONT-SIZE:13px"><span><span style="FONT-FAMILY:times new roman,new york,times,serif;COLOR:rgb(3,61,33);FONT-WEIGHT:bold;TEXT-DECORATION:underline">Cell:
09167989111</span><br></span></div>
<div> </div>
<div style="FONT-FAMILY:times new roman,new york,times,serif;FONT-SIZE:12pt">
<div style="FONT-FAMILY:times new roman,new york,times,serif;FONT-SIZE:12pt">
<div dir="ltr"><font face="Arial">
<hr size="1">
<b><span style="FONT-WEIGHT:bold">From:</span></b> Rodrigo Ferreira
<<a href="mailto:rodrigo.ferreira@vipway.net.br" target="_blank">rodrigo.ferreira@vipway.net.br</a>><br><b><span style="FONT-WEIGHT:bold">To:</span></b> OpenSIPS users mailling list
<<a href="mailto:users@lists.opensips.org" target="_blank">users@lists.opensips.org</a>> <br><b><span style="FONT-WEIGHT:bold">Sent:</span></b> Monday, 2 July 2012 10:40
PM<br><b><span style="FONT-WEIGHT:bold">Subject:</span></b>
[OpenSIPS-Users] Calls being disconnected<br></font></div>
<div>
<div>
<div> </div>
<div>
<div dir="ltr">
<div dir="ltr">
<div style="FONT-FAMILY:'Calibri';FONT-SIZE:12pt">
<div>Hey guys,</div>
<div> </div>
<div>I’m having problems with calls being “disconnected” and I dont know
where I can start to look at, because on my CDR all those calls ends
with a 200OK, but that isnt true, because you are talking all the
suddenly the call stop.</div>
<div> </div>
<div>Any ideas where I should start looking at?</div>
<div> </div>
<div style="FONT-FAMILY:'Calibri';FONT-SIZE:12pt">Engº Rodrigo
Ferreira<br>Supervisor de Telefonia<br>VIPWay Telecom<br>Tel.: +55 13
4010-1000<br>Cel.: <a href="tel:%2B55%2013%208136-5839" value="+551381365839" target="_blank">+55 13 8136-5839</a></div></div></div></div></div>
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