<font class="Apple-style-span" face="verdana, sans-serif"><div><font class="Apple-style-span" face="verdana, sans-serif">Hi All,</font></div><div><font class="Apple-style-span" face="verdana, sans-serif"><br></font></div><div>
<font class="Apple-style-span" face="verdana, sans-serif"><br></font></div>I stay away with Aastra phone for a long time and today I did some tests. All of my test ... the phones not reliable with FreeSWITCH</font><div>
<font class="Apple-style-span" face="verdana, sans-serif"><br>
</font></div><div><span class="Apple-style-span" style="font-family: verdana, sans-serif; ">I try both 6731i and 57i with the most recent firmware</span></div><div><font class="Apple-style-span" face="verdana, sans-serif"><br>
</font></div><div><font class="Apple-style-span" face="verdana, sans-serif">Here is the configuration files</font></div><div><font class="Apple-style-span" face="verdana, sans-serif"><br></font></div><div><font class="Apple-style-span" face="verdana, sans-serif"><b><u>aastra.cfg</u></b></font></div>
<div><font class="Apple-style-span" face="verdana, sans-serif"><br></font></div><div><font class="Apple-style-span" face="verdana, sans-serif"><div>dhcp: 1<span class="Apple-tab-span" style="white-space:pre"> </span></div>
<div>sip digit timeout: 3 <span class="Apple-tab-span" style="white-space:pre"> </span></div><div>sip dial plan: "x+#|xx+*|[2-9]XX[2-9]XXXXXX|1[2-9]XX[2-9]XXXXXX|1XXXXXXXXXX|[2-3]XX|67[2-9]XX[2-9]XXXXXX"</div>
<div>sip rport: 1</div><div>sip customized codec:payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=off,payload=18;ptime=20;silsupp=off</div><div>#sip registration period: 120</div><div>#sip registration renewal timer: 15</div>
<div>headset tx gain: -3</div><div>headset sidetone gain: -3</div><div>handset tx gain: -3</div><div>handset sidetone gain: -3</div><div>handsfree tx gain: 0</div><div>handset volume: 5 #RX volumes - user adjustable, so easily changed</div>
<div>speaker volume: 5</div><div>ringer volume: 3</div><div>web interface enabled: 1</div><div>live dialpad: 1 <span class="Apple-tab-span" style="white-space:pre"> </span></div><div>missed calls indicator disabled: 1<span class="Apple-tab-span" style="white-space:pre"> </span></div>
<div>suppress dtmf playback: 0<span class="Apple-tab-span" style="white-space:pre"> </span></div><div>#audio mode: 2<span class="Apple-tab-span" style="white-space:pre"> </span>#0 = speaker (default)1 = headset 2 = speaker/headset 3 = headset/speaker</div>
<div>time server disabled: 0<span class="Apple-tab-span" style="white-space:pre"> </span></div><div>time server1: <a href="http://pool.ntp.org">pool.ntp.org</a></div><div>#directory</div><div>directory 1: internal_list.csv</div>
<div>directory 2: external_list.csv</div><div><br></div><div><b><u>mac.cfg</u></b></div><div><br></div><div><div>directed call pickup: 1</div><div>directed call pickup prefix: **</div><div>#</div><div>sip line1 screen name: Ext 203</div>
<div>sip line1 display name: Ext 203</div><div>sip line1 auth name: 203</div><div>sip line1 user name: 203</div><div>sip line1 password: *********</div><div>sip line1 vmail: *97</div><div>sip line1 mode: 0</div><div>#<span class="Apple-tab-span" style="white-space:pre"> </span></div>
<div>sip line1 proxy ip: <a href="http://aastra.mydomain.com">aastra.mydomain.com</a></div><div>#sip line1 proxy port: 5060</div><div>sip line1 registrar ip: <a href="http://aastra.mydomain.com">aastra.mydomain.com</a></div>
<div>sip line1 registration period: 300</div><div>#sip line1 registrar port: 5060</div></div></font></div><div><font class="Apple-style-span" face="verdana, sans-serif"><br></font>------------ </div><div><br></div><div>1. FreeSWITCH Registration shows two entires in the internal profile and when I try to call to the extension two lines on the phones rings .... no idea why this is happen?</div>
<div><br></div><div><div>Call-ID: d8fd4c8795801cd2</div><div>User: <a href="mailto:203@aastra.mydomain.com">203@aastra.mydomain.com</a></div><div>Contact: "Ext 203" <sip:203@192.168.240.106:5060;transport=udp;fs_nat=yes;fs_path=sip%3A203%40173.33.178.49%3A1627%3Btransport%3Dudp></div>
<div>Agent: Aastra 6731i/2.6.0.2010</div><div>Status: Registered(UDP-NAT)(unknown) EXP(2011-02-16 22:24:36) EXPSECS(310)</div><div>Host: li176-12</div><div>IP: 173.33.178.49</div><div>
Port: 1627</div><div>Auth-User: 203</div><div>Auth-Realm: <a href="http://aastra.mydomain.com">aastra.mydomain.com</a></div><div>MWI-Account: <a href="mailto:203@aastra.mydomain.com">203@aastra.mydomain.com</a></div>
<div><br></div><div>Call-ID: d8fd4c8795801cd2</div><div>User: <a href="mailto:203@aastra.mydomain.com">203@aastra.mydomain.com</a></div><div>Contact: "Ext 203" <sip:203@173.33.178.49:1627;transport=udp></div>
<div>Agent: Aastra 6731i/2.6.0.2010</div><div>Status: Registered(UDP)(unknown) EXP(2011-02-16 22:24:36) EXPSECS(310)</div><div>Host: li176-12</div><div>IP: 173.33.178.49</div><div>Port: 1627</div>
<div>Auth-User: 203</div><div>Auth-Realm: <a href="http://aastra.mydomain.com">aastra.mydomain.com</a></div><div>MWI-Account: <a href="mailto:203@aastra.mydomain.com">203@aastra.mydomain.com</a></div></div><div>
<br></div><div><br>2. After first registration expires .... FreeSWITCH internal registration status shows the following entry. When I dial the extension now there is a long delay ... FreeSWITCH dialing and waiting then goes to voicemail.</div>
<div><br></div><div><div>Call-ID: d8fd4c8795801cd2</div><div>User: <a href="mailto:203@aastra.mydomain.com">203@aastra.mydomain.com</a></div><div>Contact: "Ext 203" <sip:203@173.33.178.49:1627;transport=udp></div>
<div>Agent: Aastra 6731i/2.6.0.2010</div><div>Status: Registered(UDP)(unknown) EXP(2011-02-16 22:31:25) EXPSECS(266)</div><div>Host: li176-12</div><div>IP: 173.33.178.49</div><div>Port: 1631</div>
<div>Auth-User: 203</div><div>Auth-Realm: <a href="http://aastra.mydomain.com">aastra.mydomain.com</a></div><div>MWI-Account: <a href="mailto:203@aastra.mydomain.com">203@aastra.mydomain.com</a></div><div><br>
</div><div>3. Phones Goes randomly "No Service" .... I see this problem several times in the past.</div><div><br></div><div>-------------</div><div><br></div><div>At the same time in my lab Linksys and Polycom Phones working without any issues with the default settings.</div>
<div><br></div><div>-------------</div><div><br></div><div>my question what is Aastra Doing differently?</div><div><br></div><div>Thanks</div><div>Lloyd</div><div><br></div><div class="gmail_quote">On Wed, Feb 16, 2011 at 10:53 AM, Tim St. Pierre <span dir="ltr"><<a href="mailto:fs-list@communicatefreely.net">fs-list@communicatefreely.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I have been setting my expires to 600.<br>
<div><div></div><div class="h5"><br>
Brian West wrote:<br>
> Are you setting your expires to > 300 seconds?<br>
><br>
> /b<br>
><br>
> On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:<br>
><br>
>><br>
>><br>
>> I have the same issue, around 275 phones in the field. I want the 275<br>
>> phones work with FreeSWITCH.<br>
>><br>
>><br>
>> Thanks<br>
>> Lloyd<br>
><br>
</div></div><div><div></div><div class="h5">> ------------------------------------------------------------------------<br>
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