canreinvite = yes in sip,conf ( trunk section ) ??<br>No t,t in dial command . No call recording in between , same codec should be supported by both trunk as well as extension . If trunk is iax2 and extension is sip then also asterisk will sit in media path .
<br><br><div><span class="gmail_quote">On 08/12/06, <b class="gmail_sendername">Alex Guan</b> <<a href="mailto:guan.alex@gmail.com">guan.alex@gmail.com</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;">
All,<br><br>This basic question might have been asked thousands of times....but anyways: when can Asterisk send out an re-INVITE to the line/trunk side?<br><br>It seems that the canreinvite does NOT matter for calls toward the trunk.
E.g. When I put a phone on hold, the re-INVITE is sent from phone to the Asterisk, but then that's it. The Asterisk never sends it out. It seems to work for extension to extention, but not extension to line. What am I missing?
<br><br>Thanks!<br>
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