Re: [OpenSER-Users] Calls disconnect automatically

On Monday 21 January 2008 00:52:24 VoIP Forums www.Go4Calls.com wrote:

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.

Try the "tcpdump" I suggested in client side, you will discover when audio is cut.