hi<br>in sip.conf there is a parameter calllimit or something like that use it...<br>David<br><br><div class="gmail_quote">2008/12/5 James Lamanna <span dir="ltr"><<a href="mailto:jlamanna@gmail.com">jlamanna@gmail.com</a>></span><br>
<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>
I've noticed that if I have a multi-line linksys (942 or 962) phone<br>
with the same sip registration mapped to each line key, that if all<br>
the lines are full the phone will accept another call. I would expect<br>
the phone to respond with "busy" so the call would to directly to<br>
voicemail.<br>
<br>
Has anyone else experienced this and know of a workaround? I know it<br>
seems like an endpoint issue and not an asterisk one.<br>
<br>
Thanks.<br>
<br>
--James<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br> (\__/) <br>(='.'=)This is Bunny. Copy and paste bunny into your <br>(")_(")signature to help him gain world domination. <br><br>