[asterisk-users] RE: RTPTIMEOUT Configuration

[itn at itn.com.br]

Hello,
 
I see the following descriptions on 
HYPERLINK
"http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+rtptimeout"rt
ptimeout = Number : Number of seconds, to wait for RTP traffic before
classify the connection as discontinued. Default 0 (no RTP timeout).
(New in v1.2.x). 
HYPERLINK
"http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf"h
ttp://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf 
 
HYPERLINK
"http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+rtptimeout"ht
tp://www.voip-info.org/wiki/index.php?page=Asterisk+sip+rtptimeout 
 
Valid only in [general] section and type=peer
 
My question is, 
 
How can I set this variable RTPTIMEOUT for a given extension terminate
the call after a pre-defined time of no RTP traffic
 
I am trying to use like
 
[general]
 
rtptimeout=30
 
 
[1770]
type=peer
username=1770
secret=desrrr
nat=yes
context=brasil
host=dynamic
dtmfmode=rfc2833
rtptimeout=30 
disallow=all
allow=g729
allow=g723
allow=ulaw
allow=ilbc
 
[1771]
type=peer
username=1771
secret=alkmvcccc
nat=yes
context=brasil
host=dynamic
dtmfmode=rfc2833
rtptimeout=30 
disallow=all
allow=g729
allow=g723
allow=ulaw
allow=ilbc
 
 
In this case, how can I set calls to terminate on extensions 1770 and
1771 after 30 seconds of no RTP traffic ???
Atenciosamente,
______________________________________________________________________
Uniglobo Telecomunicacoes Ltda
 Diretoria Comercial - Newton Medina
PBX 55-11-4082-3555
FAX 55-11-4581-9659
MSN itn at itn.com.br 
VISITE HYPERLINK
"file:///C:\\Documents%20and%20Settings\\Uniglobo%20Telecom\\Application
%20Data\\Microsoft\\Signatures\\www.uniglobo.com.br"www.uniglobo.com.br 
 

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