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 [EMAIL PROTECTED] 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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