21 Jan
2008
21 Jan
'08
10:41 a.m.
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.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es