<font class="Apple-style-span" face="verdana, sans-serif"><br></font><div><font face="verdana,sans-serif"><br></font></div><div><font face="verdana,sans-serif">I have the same issue, around 275 phones in the field. I want the 275 phones work with FreeSWITCH.</font></div>

<div><font face="verdana,sans-serif"><br></font></div><div><font face="verdana,sans-serif"><br></font></div><div><font face="verdana,sans-serif">Thanks<br clear="all"></font><font class="Apple-style-span" face="verdana, sans-serif">Lloyd<br>

</font>
<br><br><div class="gmail_quote">On Tue, Feb 15, 2011 at 10:29 AM, Tim St. Pierre <span dir="ltr">&lt;<a href="mailto:fs-list@communicatefreely.net">fs-list@communicatefreely.net</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

Good luck!<br>
<br>
There are some aastra.cfg sections earlier in this thread.  One thing I<br>
found about the 9133i and other legacy phones is that they work better<br>
if you don&#39;t subscribe to MWI but instead send unsolicited updates.<br>
They also have odd default codec settings, so you want to specify the<br>
codec string with the ptime that you want (I think they default to 30<br>
instead of 20).<br>
<br>
I like the newer phones much better, but we have about 150 of these in<br>
the field, so we have to make them work.<br>
<div class="im"><br>
-Tim<br>
<br>
Aloysius Lloyd wrote:<br>
&gt; Tim,<br>
&gt;<br>
</div><div class="im">&gt; Thank you for the settings will give a try.<br>
&gt;<br>
&gt;<br>
&gt; Thanks<br>
&gt; Lloyd<br>
&gt;<br>
&gt; On Mon, Feb 14, 2011 at 6:53 PM, Tim St. Pierre<br>
</div><div class="im">&gt; &lt;<a href="mailto:fs-list@communicatefreely.net">fs-list@communicatefreely.net</a> &lt;mailto:<a href="mailto:fs-list@communicatefreely.net">fs-list@communicatefreely.net</a>&gt;&gt;<br>
&gt; wrote:<br>
&gt;<br>
</div><div class="im">&gt;     &lt;domains&gt;<br>
&gt;        &lt;domain name=&quot;all&quot; alias=&quot;true&quot; parse=&quot;true&quot;/&gt;<br>
</div>&gt;        &lt;domain name=&quot;<a href="http://pbx.MYDOMAIN.NET" target="_blank">pbx.MYDOMAIN.NET</a> &lt;<a href="http://pbx.MYDOMAIN.NET" target="_blank">http://pbx.MYDOMAIN.NET</a>&gt;&quot;<br>
<div><div></div><div class="h5">&gt;     alias=&quot;true&quot; parse=&quot;false&quot;/&gt;<br>
&gt;      &lt;/domains&gt;<br>
&gt;<br>
&gt;     &lt;settings&gt;<br>
&gt;        &lt;param name=&quot;user-agent-string&quot; value=&quot;Communicate Freely 2.0&quot;/&gt;<br>
&gt;        &lt;param name=&quot;debug&quot; value=&quot;0&quot;/&gt;<br>
&gt;        &lt;param name=&quot;sip-trace&quot; value=&quot;no&quot;/&gt;<br>
&gt;        &lt;param name=&quot;log-auth-failures&quot; value=&quot;true&quot;/&gt;<br>
&gt;        &lt;param name=&quot;rfc2833-pt&quot; value=&quot;101&quot;/&gt;<br>
&gt;        &lt;param name=&quot;sip-port&quot; value=&quot;$${internal_sip_port}&quot;/&gt;<br>
&gt;        &lt;param name=&quot;dialplan&quot; value=&quot;xml&quot;/&gt;<br>
&gt;        &lt;param name=&quot;context&quot; value=&quot;internal&quot;/&gt;<br>
&gt;        &lt;param name=&quot;dtmf-duration&quot; value=&quot;100&quot;/&gt;<br>
&gt;        &lt;param name=&quot;inbound-codec-prefs&quot;<br>
&gt;     value=&quot;$${internal_codec_prefs}&quot;/&gt;<br>
&gt;        &lt;param name=&quot;outbound-codec-prefs&quot;<br>
&gt;     value=&quot;$${internal_codec_prefs}&quot;/&gt;<br>
&gt;        &lt;param name=&quot;rtp-timer-name&quot; value=&quot;soft&quot;/&gt;<br>
&gt;        &lt;param name=&quot;sip-ip&quot; value=&quot;$${public_ip}&quot;/&gt;<br>
&gt;        &lt;param name=&quot;rtp-ip&quot; value=&quot;$${public_ip}&quot;/&gt;<br>
&gt;        &lt;param name=&quot;hold-music&quot; value=&quot;$${moh_prefix}alt&quot;/&gt;<br>
&gt;        &lt;param name=&quot;dtmf-type&quot; value=&quot;rfc2833&quot;/&gt;<br>
&gt;<br>
&gt;        &lt;param name=&quot;force-register-domain&quot; value=&quot;$${domain}&quot;/&gt;<br>
&gt;        &lt;param name=&quot;force-subscription-domain&quot; value=&quot;pbx.$${domain}&quot;/&gt;<br>
&gt;        &lt;param name=&quot;force-register-db-domain&quot; value=&quot;$${domain}&quot;/&gt;<br>
&gt;        &lt;param name=&quot;force-subscription-expires&quot; value=&quot;600&quot;/&gt;<br>
&gt;<br>
&gt;     These are the most important ones I think.<br>
&gt;<br>
&gt;        &lt;param name=&quot;NDLB-received-in-nat-reg-contact&quot; value=&quot;true&quot;/&gt;<br>
&gt;        &lt;param name=&quot;sip-force-contact&quot; value=&quot;<br>
&gt;     NDLB-connectile-dysfunction&quot;/&gt;<br>
&gt;<br>
&gt;     I&#39;m also using sip-force-expires to 600 at the moment, and ping =<br>
&gt;     10.  I<br>
&gt;     will probably increase those eventually to reduce bandwidth.  I&#39;m<br>
&gt;     still<br>
&gt;     in Beta right now, but I&#39;m not having too many issues.<br>
&gt;<br>
&gt;     Some of those params are added per-device using the directory, so<br>
&gt;     I can<br>
&gt;     tweak them depending on which device registers, and what the NAT<br>
&gt;     status<br>
&gt;     is of that device.  I&#39;m really pushing to get IPv6 on the phones, as<br>
&gt;     well as on some of the more prominent (but competitive) DSL providers<br>
&gt;     here so that we can forego NAT altogether some day.<br>
&gt;<br>
&gt;     Hope that&#39;s helpful.  I haven&#39;t really gone through and figured out<br>
&gt;     which variables do what at the moment, but it seems to work as it is.<br>
&gt;<br>
&gt;     -Tim<br>
&gt;     Aloysius Lloyd wrote:<br>
&gt;     &gt; Tim,<br>
&gt;     &gt;<br>
&gt;     &gt; Thank you for the information.<br>
&gt;     &gt;<br>
&gt;     &gt; I have around 275 Aastra 9133i models phones in production .These<br>
&gt;     &gt; phones installed 31/2 years ago. I am trying to migrate to<br>
&gt;     FreeSWITCH<br>
&gt;     &gt; could not make it work reliably.<br>
&gt;     &gt;<br>
&gt;     &gt; What are the profile settings turned on for these phones works<br>
&gt;     reliably?<br>
&gt;     &gt;<br>
&gt;     &gt;<br>
&gt;     &gt; Thanks<br>
&gt;     &gt; Lloyd<br>
&gt;     &gt;<br>
&gt;     &gt;<br>
&gt;     &gt;<br>
&gt;     &gt; On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre<br>
&gt;     &gt; &lt;<a href="mailto:fs-list@communicatefreely.net">fs-list@communicatefreely.net</a><br>
&gt;     &lt;mailto:<a href="mailto:fs-list@communicatefreely.net">fs-list@communicatefreely.net</a>&gt;<br>
&gt;     &lt;mailto:<a href="mailto:fs-list@communicatefreely.net">fs-list@communicatefreely.net</a><br>
&gt;     &lt;mailto:<a href="mailto:fs-list@communicatefreely.net">fs-list@communicatefreely.net</a>&gt;&gt;&gt;<br>
&gt;     &gt; wrote:<br>
&gt;     &gt;<br>
&gt;     &gt;     Sure,<br>
&gt;     &gt;<br>
&gt;     &gt;     I normally administer about 300 Aastra phones, with every<br>
&gt;     model they<br>
&gt;     &gt;     make represented.<br>
&gt;     &gt;<br>
&gt;     &gt;     I have 22 connected to our Freeswitch &quot;beta&quot; system, which will<br>
&gt;     &gt;     eventually become production.<br>
&gt;     &gt;<br>
&gt;     &gt;     All the endpoints are behind NAT without exception.  There are a<br>
&gt;     &gt;     number<br>
&gt;     &gt;     of legacy 9133i and 480i phones on the network that don&#39;t<br>
&gt;     have the<br>
&gt;     &gt;     newer<br>
&gt;     &gt;     NAT traversal features available, but this doesn&#39;t seem to be a<br>
&gt;     &gt;     problem.  I have some of the nat traversal options turned on<br>
&gt;     in the<br>
&gt;     &gt;     sofia profile though, so fs will send media back to the<br>
&gt;     originating<br>
&gt;     &gt;     address and port.<br>
&gt;     &gt;<br>
&gt;     &gt;     They have been quite reliable, and the sound quality has been<br>
&gt;     &gt;     excellent,<br>
&gt;     &gt;     with the newer phones using g722 at 16KHz.<br>
&gt;     &gt;<br>
&gt;     &gt;     There are a few advanced features that I haven&#39;t had a<br>
&gt;     chance to play<br>
&gt;     &gt;     with yet, but here&#39;s what I have working:<br>
&gt;     &gt;<br>
&gt;     &gt;     Regular calls, in and out.<br>
&gt;     &gt;     Intercom calls (auto-answer to speaker phone)<br>
&gt;     &gt;     Automatic update of destination name and number (updates<br>
&gt;     when checking<br>
&gt;     &gt;     voice mail, and when calling an extension).  Only on newer<br>
&gt;     phones<br>
&gt;     &gt;     Blind and attended transfer<br>
&gt;     &gt;     Music on hold<br>
&gt;     &gt;     SIP using udp or tcp (haven&#39;t tried TLS yet)<br>
&gt;     &gt;     Fewer issues with DTMF than with asterisk, using rfc2833<br>
&gt;     dtmf (no<br>
&gt;     &gt;     issues<br>
&gt;     &gt;     as of yet).<br>
&gt;     &gt;     BLF lamps work correctly, flashing when the phone rings, lit<br>
&gt;     &gt;     steady when<br>
&gt;     &gt;     they are on the phone.<br>
&gt;     &gt;     Distinctive ringing works.<br>
&gt;     &gt;     I haven&#39;t tried SLA yet, but Aastra recently released a firmware<br>
&gt;     &gt;     update<br>
&gt;     &gt;     that fixes a missing header, reported to have broken correct SLA<br>
&gt;     &gt;     operation.  I&#39;m hoping to test that in the next week or two.<br>
&gt;     &gt;<br>
&gt;     &gt;     The phones provision very nicely - we auto generate config<br>
&gt;     using PHP<br>
&gt;     &gt;     scripts that generate a config file on the fly from the user<br>
&gt;     database.<br>
&gt;     &gt;     These are very easy phones to deploy in large installations,<br>
&gt;     or to the<br>
&gt;     &gt;     outside world (not readily accessible).  They have just<br>
&gt;     added some new<br>
&gt;     &gt;     features that allow for remote diagnostics of the phones as<br>
&gt;     well.<br>
&gt;     &gt;<br>
&gt;     &gt;     There is a great deal of XML programmability in the phones<br>
&gt;     too, which<br>
&gt;     &gt;     I&#39;m starting to use for call control and other useful things<br>
&gt;     (updating<br>
&gt;     &gt;     forwarding rules in the database, or conference and<br>
&gt;     recording control<br>
&gt;     &gt;     using ESL).<br>
&gt;     &gt;<br>
&gt;     &gt;     Hope that helps!<br>
&gt;     &gt;<br>
&gt;     &gt;     -Tim<br>
&gt;     &gt;<br>
&gt;     &gt;     Aloysius Lloyd wrote:<br>
&gt;     &gt;     &gt; Tim,<br>
&gt;     &gt;     &gt;<br>
&gt;     &gt;     &gt; Can you share your success stories FreeSWITCH and Aastra.<br>
&gt;     &gt;     &gt;<br>
&gt;     &gt;     &gt; Aastra Phones Behind the NAT?<br>
&gt;     &gt;     &gt;<br>
&gt;     &gt;     &gt; In my case Aastra phones registration not a problem.<br>
&gt;     &gt;     &gt;<br>
&gt;     &gt;     &gt; But calls drooped every 60 sec ... in the same environment<br>
&gt;     &gt;     Linksys and<br>
&gt;     &gt;     &gt; Polycom works perfectly.<br>
&gt;     &gt;     &gt;<br>
&gt;     &gt;     &gt; How stable the Aastra phones with FreeSWITCH system.<br>
&gt;     &gt;     &gt;<br>
&gt;     &gt;     &gt; TIA<br>
&gt;     &gt;     &gt;<br>
&gt;     &gt;     &gt; Lloyd<br>
&gt;     &gt;<br>
&gt;     &gt;<br>
&gt;     &gt;     _______________________________________________<br>
&gt;     &gt;     FreeSWITCH-users mailing list<br>
&gt;     &gt;     <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
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&gt;     &gt;<br>
&gt;     &gt;<br>
&gt;     &gt;<br>
&gt;     ------------------------------------------------------------------------<br>
&gt;     &gt;<br>
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&gt;<br>
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</div></div></blockquote></div><br></div>