[OpenSER-Users] Calls disconnect automatically

Marc LEURENT lftsy at free.fr
Mon Jan 21 10:05:13 CET 2008


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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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