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<DIV>That was a typo its corrected to [8407] but problem still persist with original questions though.</DIV>
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<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">-------------- Original message -------------- <BR>From: "Eric "ManxPower" Wieling" <eric@fnords.org> <BR><BR>> "[9507]" is the incoming User ID. "user=8407" is the outgoing User ID. <BR>> Do you really want them to be different? <BR>> <BR>> Dial() will stop processing of the dialplan until the call ends. Do you <BR>> really want this? <BR>> <BR>> "r" option to Dial will force a ringing sound to the caller, even if the <BR>> caller should be hearing a "all circuits are busy", or "your call cannot <BR>> be completed as dialed" or similar message. Do you really want that? <BR>> <BR>> broadbandvoice@comcast.net wrote: <BR>> > Thanks for the response, its looks logical, for some reason the authentication <BR>> is not working for me, I'm using a Linksys Phone adapter and here is a sample <BR>> dial plan in extensions.conf and also SIP channels. <BR>&
gt; > <BR>> > exten => 8407,1,Dial(SIP/8407,80,rt) ; permit transfer <BR>> > exten => 8407,n,Authenticate(9461) <BR>> > exten => 8407,n,Playback(pbx-invalid) <BR>> > exten => 8407,n,Hangup() <BR>> > <BR>> > and in sip.conf <BR>> > <BR>> > [9507] <BR>> > type=friend <BR>> > user=8407 <BR>> > secret=xxxxxxxxxx <BR>> > ;context=from-sip <BR>> > callerid=8407 <BR>> > host=dynamic <BR>> > nat=yes <BR>> > qualify=yes <BR>> > canreinvite=no <BR>> > dtmfmode=rfc2833 <BR>> <BR>> -- <BR>> Now accepting new clients in Birmingham, Atlanta, Huntsville, <BR>> Chattanooga, and Montgomery. <BR>> _______________________________________________ <BR>> --Bandwidth and Colocation provided by Easynews.com -- <BR>> <BR>> asterisk-users mailing list <BR>> To UNSUBSCRIBE or update options visit: <BR>> http://lists.digium.com/mailman/listinfo/asterisk-users
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