Hi,<br>
Who is giving you that notice. If I undrestand wright, you are using
SER as sip proxy and asterisk as media gateway. What adapters (PSTN,
ISDN, GSM ...) are you using on asterisk? Did you try to make two
concurent calls with users on asterisk? <br>
Do you have tihis line in extensions.conf :
exten => xxx,xxx,Hangup<br>
How did you configure your adapter, depending on it's type?<br>
<br><br><div><span class="gmail_quote">On 4/20/06, <b class="gmail_sendername">fay butt</b> <<a href="mailto:fay_butt@yahoo.com">fay_butt@yahoo.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;">
<div style="direction: ltr;">Hi Ivan,<br>
I tried with the same configuration procedure on
different box's. but when i dials it makes call (first time but with
delay) , In the second attempt it shows the messege
"Everyone is busy/congested at this time (1:0/1/0)" <br> "Auto fallthrough channel 'SIP/phone1-9022' status is 'CONGESTION' <br> I am surprised, what's wrong .<br> Plz acknowledge,<br> m.F.B
</div><div style="direction: ltr;"><span class="e" id="q_10ab78d67b1ba9d0_1"><br> <b><i><br>
</i></b></span></div></blockquote></div>