<HTML><HEAD></HEAD>
<BODY dir=ltr>
<DIV dir=ltr>
<DIV style="FONT-FAMILY: 'Calibri'; COLOR: #000000; FONT-SIZE: 12pt">
<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-STYLE: normal; DISPLAY: inline; FONT-FAMILY: 'Calibri'; COLOR: #000000; FONT-SIZE: small; FONT-WEIGHT: normal; TEXT-DECORATION: none">
<DIV style="FONT: 10pt tahoma">
<DIV> </DIV>
<DIV style="BACKGROUND: #f5f5f5">
<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> Wednesday, July 04, 2012 1:54 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">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 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>
<DIV class=h5>
<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" target=_blank
value="+551381365839">+55 13 8136-5839</A></DIV></DIV></DIV></DIV></DIV>
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