I’m attaching the log that I got from a call that was disconnected

From: Rodrigo Ferreira 
Sent: Tuesday, July 03, 2012 11:09 AM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Calls being disconnected

I’m getting the logs, but let me ask you something, force_rport will help me? 

From: SamyGo 
Sent: Tuesday, July 03, 2012 8:18 AM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Calls being disconnected

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. 


On Tue, Jul 3, 2012 at 4:12 PM, Rodrigo Ferreira 
<[email protected]> wrote:

  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.

  From: SamyGo 
  Sent: Tuesday, July 03, 2012 8:02 AM
  To: OpenSIPS users mailling list 
  Subject: Re: [OpenSIPS-Users] Calls being disconnected

  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 !!


  On Tue, Jul 3, 2012 at 3:58 PM, Rodrigo Ferreira 
<[email protected]> wrote:

    I will take a look at my rtp .. I’m using rtpproxy on my box, I will try to 
get some logs today too

    From: SamyGo 
    Sent: Tuesday, July 03, 2012 4:34 AM
    To: aamir chougule ; OpenSIPS users mailling list 
    Subject: Re: [OpenSIPS-Users] Calls being disconnected

    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 
directly.

    Regards, 
    Sammy

    On Tue, Jul 3, 2012 at 12:21 PM, aamir chougule <[email protected]> wrote:

      Hi Rodrigo,


      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.


      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.

      Regards,

      Aamir Chougule
      Cell: 09167989111



--------------------------------------------------------------------------
      From: Rodrigo Ferreira <[email protected]>
      To: OpenSIPS users mailling list <[email protected]> 
      Sent: Monday, 2 July 2012 10:40 PM
      Subject: [OpenSIPS-Users] Calls being disconnected


      Hey guys,

      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.

      Any ideas where I should start looking at?

      Engº Rodrigo Ferreira
      Supervisor de Telefonia
      VIPWay Telecom
      Tel.: +55 13 4010-1000
      Cel.: +55 13 8136-5839

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