The problem is only with PSTN call.
 
I tried to send call to the three gateway Teles, SIP-HIT and Asterisk but all disconnect calls in that priticular seconds.
The thinng is i cannot understand if i am using STUN in Linksyspap2 the call goes normal and without STUN it disconnect. So the problem is gateway side or Openser?
 
our router is not implimented with SIP, and there is one more strange thing, In some callshop the same rtptproxy working well and going cal for long duration but i have 3 callshop which facing this problem. the configuration and others are same as other working devices.
 
 
 


Regards,
www.Go4Calls.Com
VoIP Forums

> From: ibc@aliax.net
> To: users@lists.openser.org
> Date: Mon, 21 Jan 2008 00:19:12 +0100
> Subject: Re: [OpenSER-Users] Calls disconnect automatically
>
> El Lunes, 21 de Enero de 2008, VoIP Forums www.Go4Calls.com escribió:
> > i tired with the following configuration but still result is same. calls
> > disconnect in 30 - 32 sec
> >
> > modparam("nathelper", "natping_interval", 20)
> > modparam("nathelper", "ping_nated_only", 1)
> > modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")
> > modparam("nathelper", "rtpproxy_disable", 0)
> > modparam("nathelper", "rtpproxy_disable_tout", 60)
> > modparam("nathelper", "rtpproxy_tout", 1)
> > modparam("nathelper", "rtpproxy_retr", 5)
> > modparam("nathelper", "sipping_method", "OPTIONS")
> > modparam("nathelper", "received_avp", "$avp(i:801)")
> >
> > Please advise me if i need more modification?
>
>
> Which kind of calls are disconnected after 30 seconds? PSTN calls or user to
> user call?
>
> In any case, you could do a "tcpdump -n port UAC_RTP_PORT" in a PC using a
> softphone that uses UAC_RTP_PORT for audio. Call to PSTN (or other user) from
> this softphone and monitorize with tcpdump when the audio is disconnected.
>
> Some gateways (as Asterisk) disconnect a call by default if they don't receive
> RTP during 30 seconds.
>
> Since I don't know which kind of gateway you use I don't know if it uses
> Session Timers as call monitorization way, so if your router blocks the port
> after 30 seconds, then the periodic ire-INVITE or UPDATE from gateway to UAC
> will not arrive so they won't be replied with "200 OK", and gateway will
> discconect the call.
> To test this, do a "ngrep" in a computer using a softphone registered behind
> NAT (no STUN). After REGISTER you should receive a OPTIONS from proxy as keep
> alive.
>
> Another possible problem is the existence of painful ALG routers, have you
> tested if your router implements SIP ALG?
>
>
>
>
> --
> Iñaki Baz Castillo
>
> _______________________________________________
> Users mailing list
> Users@lists.openser.org
> http://lists.openser.org/cgi-bin/mailman/listinfo/users

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