Hi All,
I have captured the packet when the device sent ICMP to the media gateway and the reply comming "DESTINATION PORT UNREACHABE" that time device send BYE message to Openser. I am not so expert to analysis the packet if you could have a look to the following URL i put already the shot of packet.
http://www.emafone.net/packet.JPG
Regards, www.Go4Calls.Com VoIP Forums
Date: Mon, 21 Jan 2008 10:05:13 +0100 From: lftsy@free.fr To: ibc@in.ilimit.es CC: users@lists.openser.org Subject: Re: [OpenSER-Users] Calls disconnect automatically
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I suppose you have a trouble with ACK paquets, I think that's why you've been disconnected after 30s... There are some clients that shutdown the call after not receiving ACK paquet after their 200OK.
As Iñaki said use tcpdump / or ngrep -W byline port 5060
to see what's wrong
Have a nice day!
Iñaki Baz Castillo a écrit :
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.
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iD8DBQFHlGBIN4+o+2LtdFwRAq4rAJ4l6N2JVEuPQB6FYkfyOsqcQlPbXwCeIuO3 9pbYaQyUTLh6M+6TSjM5kH0= =haPm -----END PGP SIGNATURE-----
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