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<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>
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<DIV style="font-color: black"><B>From:</B> <A title=govoiper@gmail.com
href="mailto:govoiper@gmail.com">SamyGo</A> </DIV>
<DIV><B>Sent:</B> Tuesday, July 03, 2012 8:02 AM</DIV>
<DIV><B>To:</B> <A title=users@lists.opensips.org
href="mailto:users@lists.opensips.org">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'; COLOR: #000000; 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>
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<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>
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<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>
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style="FONT-STYLE: normal; DISPLAY: inline; FONT-FAMILY: 'Calibri'; FONT-SIZE: small; FONT-WEIGHT: normal; TEXT-DECORATION: none">
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<DIV class=h5>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>
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<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>
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<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" target=_blank
value="+551381365839">+55 13 8136-5839</A></DIV></DIV></DIV></DIV></DIV>
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<P align=left color="#000000">Nenhum vírus encontrado nessa
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<P align=left color="#000000" avgcert??>Nenhum vírus encontrado nessa
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