When Dialing via SIP provider you don't have the concept of lines.<br><br>Your operator probably allow you more than one simultaneous call at the same time.<br>So concerning with your diaplan on extensions.conf, only if the first dial execution couldn't be reached asterisk will lookup for the 2nd priority. It doesn't have any relation ship with number of lines...
<br><br>Please check on voip wiki about SIP trunking , SIP peers to understand the concept.<br><br><br><br><div><span class="gmail_quote">On 11/20/06, <b class="gmail_sendername">Noc Phibee</b> <<a href="mailto:noc@phibee.net">
noc@phibee.net</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi<br><br>no answer ? anyone use asterisk with more 1 line in out ?
<br><br>bye<br><br><br><br>Noc Phibee a écrit :<br>> Hi<br>><br>> actually, for out call, i use :<br>><br>><br>> exten => _0XXXX.,1,Dial(SIP/out-l1/${EXTEN:1},50,rt)<br>> exten => _0XXXX.,2,Dial(SIP/out-l2/${EXTEN:1},50,rt)
<br>> exten => _0XXXX.,3,Dial(SIP/out-l3/${EXTEN:1},50,rt)<br>> exten => _0XXXX.,4,Hangup<br>><br>><br>> can you say me with this config, if the first user call and use out-l1<br>> the second user use automatiquely out-l2 (and out-l3 when l1 and l2 are
<br>> used) ?<br>><br>> if i want add a "turn line" for use all lines and not only the out-l1,<br>> what is the best<br>> config (i don't use web interface for config)<br>><br>> thanks bye<br>
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