<br><br><div><span class="gmail_quote">2006/6/22, Michiel van Baak <<a href="mailto:michiel@vanbaak.info">michiel@vanbaak.info</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
On 16:57, Thu 22 Jun 06, Benjamin Stocker wrote:<br>> Hi<br>><br>> This works fine in extensions.conf:<br>><br>> exten => _0X./100,1,Dial(SIP/${EXTEN}@sipout-a)<br>> exten => _0X./200,1,Dial(SIP/${EXTEN}@sipout-a
)<br>><br>> This will just use different SIP channels for different Caller ID's.<br>> If I write the same to a realtime table, Asterisk always uses sipout-a, no<br>> matter what Caller ID is used.<br><br>That will be the case with static configs too, because the
<br>argument to Dial is the same in both cases<br><br></blockquote></div><br>That was a typo. Sorry, the second line reads:<br><br>exten => _0X./200,1,Dial(SIP/${EXTEN}@sipout-b)<br>