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Have you set a call limit for each SIP peer?&nbsp; This is now required as
of version 1.4.&nbsp; It took me a while to figure out all the issues when
migrating to 1.4.<br>
<br>
<br>
John Meksavan wrote:
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  </style>Thanks for you guys help.&nbsp; The status LED&nbsp; on the sidecar
takes an awfully look time to change from GREEN to RED and vice versa.&nbsp;
Some times, it would reguire up to 15-20 minutes at beginning or ending
the call on the extension. <br>
  <br>
What would cause the delay?&nbsp; Is it my network?<br>
  <br>
Best Regards,<br>
John<br>
  <br>
&gt; Date: Fri, 28 Mar 2008 09:05:02 -0600<br>
&gt; From: <a class="moz-txt-link-abbreviated" href="mailto:sean@datawhale.com">sean@datawhale.com</a><br>
&gt; To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
&gt; Subject: Re: [asterisk-users] SPA-962+ SPA-932- blf function<br>
&gt; <br>
&gt; John Meksavan wrote:<br>
&gt; &gt; Asterisk Users,<br>
&gt; &gt;<br>
&gt; &gt; I am running Asterisk 1.4.11 on Debian "Etch" system with the
TDM03B <br>
&gt; &gt; wildcard. I recently purchased a SPA-962 and SPA-932- the
sidecar for <br>
&gt; &gt; our receptionist. After reading many forum postings on how to
  <br>
&gt; &gt; configure the side car, I uprgraded the SPA-962 software to <br>
&gt; &gt; 5.1.18(SC) version. <br>
&gt; &gt;<br>
&gt; &gt; I got the sidecar to subscribed to an extension on the
Asterisk <br>
&gt; &gt; server, but the LED state on the SPA-932 never changes even
when I am <br>
&gt; &gt; a call with that extension on another VOIP phone- SPA-941. I
got the <br>
&gt; &gt; speed dial function to work, but the "blf" function does not
appear to <br>
&gt; &gt; work. <br>
&gt; &gt;<br>
&gt; &gt; Did anybody get the "blf" function to work? What I am doing
wrong? <br>
&gt; &gt; Any input would be greatly appreciated. Thanks in advance. <br>
&gt; &gt;<br>
&gt; &gt; Regards,<br>
&gt; &gt; John<br>
&gt; &gt;
------------------------------------------------------------------------<br>
&gt; &gt; How well do you know your celebrity gossip? Talk celebrity
smackdowns <br>
&gt; &gt; here.
<a class="moz-txt-link-rfc2396E" href="http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A">&lt;http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A&gt;</a><br>
&gt; &gt;
------------------------------------------------------------------------<br>
&gt; &gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; -- Bandwidth and Colocation Provided by
<a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>
&gt; &gt;<br>
&gt; &gt; asterisk-users mailing list<br>
&gt; &gt; To UNSUBSCRIBE or update options visit:<br>
&gt; &gt; <a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
&gt; To make it work properly I had to add the following to sip.conf:<br>
&gt; allowsubscribe=yes<br>
&gt; notifyringing=yes<br>
&gt; limitonpeer=yes<br>
&gt; notifyhold=yes<br>
&gt; <br>
&gt; See if that helps.<br>
&gt; <br>
&gt; -Sean<br>
&gt; <br>
&gt; <br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; -- Bandwidth and Colocation Provided by <a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a>
--<br>
&gt; <br>
&gt; asterisk-users mailing list<br>
&gt; To UNSUBSCRIBE or update options visit:<br>
&gt; <a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
  <br>
  <hr>How well do you know your celebrity gossip? <a
 moz-do-not-send="true"
 href="http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A"
 target="_new">Talk celebrity smackdowns here.</a>
  <pre wrap="">
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