[OpenSIPS-Users] Calls being disconnected
Rodrigo Ferreira
rodrigo.ferreira at vipway.net.br
Tue Jul 3 16:14:10 CEST 2012
I already have force_rport on my script, nevermind
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 <rodrigo.ferreira at vipway.net.br> 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 <rodrigo.ferreira at vipway.net.br> 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 <aamir_ryu at yahoo.com> 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 <rodrigo.ferreira at vipway.net.br>
To: OpenSIPS users mailling list <users at lists.opensips.org>
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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