Get the SIP trace of the phone doing its registration. Also, is the phone on the same LAN as FreeSWITCH? Is there any NAT involved?<div>-MC<br><br><div class="gmail_quote">On Fri, Feb 18, 2011 at 12:43 PM, Aloysius Lloyd <span dir="ltr"><<a href="mailto:lloyd.aloysius@gmail.com">lloyd.aloysius@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><font face="verdana, sans-serif">I did some traces and figure it out the problem losing the Registrations.</font><div>
<font face="verdana, sans-serif"><br></font></div><div>
<font face="verdana, sans-serif">1. FreeSWITCH Send the response to a wrong port . Is this a Bug?</font></div><div><font face="verdana, sans-serif"><br></font></div><div>
<font face="verdana, sans-serif">2. Why FreeSWITCH + Aastra Registration contact not have the port number ?</font></div>
<div><font face="verdana, sans-serif"><br></font></div><div><font face="verdana, sans-serif">3. See the below ... </font></div><div><font face="verdana, sans-serif"><br>
</font></div><div><font face="verdana, sans-serif">203 & 202 Aastra Phones on Domain - </font><span style="font-family:verdana, sans-serif"><b><a href="http://aastra.mydomain.com" target="_blank">aastra.mydomain.com</a></b></span></div>
<div><font face="verdana, sans-serif"><br></font></div><div><font face="verdana, sans-serif">But user Agent Shows Polycom. BTW in the same LAN I have two polycom phones on different domain <b><a href="http://foo.mydomain.com" target="_blank">foo.mydomain.com</a></b></font></div>
<div><font face="verdana, sans-serif"><br></font></div><div><font face="verdana, sans-serif">==============</font></div><div><font face="verdana, sans-serif"><br>
</font></div><div><font face="verdana, sans-serif"><div>SIP/2.0 200 OK</div><div>Via: SIP/2.0/UDP 173.230.136.12;rport;branch=z9hG4bKZ6DDSDNZ11gvS</div><div><b>From: <<a href="mailto:sip%3A203@aastra.mydomain.com" target="_blank">sip:203@aastra.mydomain.com</a>>;tag=3jFNS2KBvNX0S</b></div>
<div><b>To: "Ext 202" <<a href="mailto:sip%3A203@aastra.mydomain.com" target="_blank">sip:203@aastra.mydomain.com</a>>;tag=B550915E-F491D63B</b></div><div>CSeq: 8639912 NOTIFY</div><div>Call-ID: cdad5068-b542-122e-ef86-fefdade6880c</div>
<div><b>Contact: <<a href="http://sip:202@192.168.240.128:5060" target="_blank">sip:202@192.168.240.128:5060</a>></b></div><div>Event: message-summary</div><div><b>User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.3.1.0769</b></div>
<div>
Accept-Language: en</div><div>Content-Length: 0</div><div><br></div><div>=============</div><div><br></div><div>Any help on this why sending wrong ports?</div></font></div><div><font face="verdana, sans-serif"><br>
</font>Thanks</div><div>Lloyd<div><div></div><div class="h5"><br><div class="gmail_quote">On Wed, Feb 16, 2011 at 10:34 PM, Aloysius Lloyd <span dir="ltr"><<a href="mailto:lloyd.aloysius@gmail.com" target="_blank">lloyd.aloysius@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<font face="verdana, sans-serif"><div><font face="verdana, sans-serif">Hi All,</font></div><div><font face="verdana, sans-serif"><br></font></div><div>
<font 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 face="verdana, sans-serif"><br>
</font></div><div><span style="font-family:verdana, sans-serif">I try both 6731i and 57i with the most recent firmware</span></div><div><font face="verdana, sans-serif"><br>
</font></div><div><font face="verdana, sans-serif">Here is the configuration files</font></div><div><font face="verdana, sans-serif"><br></font></div><div><font face="verdana, sans-serif"><b><u>aastra.cfg</u></b></font></div>
<div><font face="verdana, sans-serif"><br></font></div><div><font face="verdana, sans-serif"><div>dhcp: 1<span style="white-space:pre-wrap"> </span></div>
<div>sip digit timeout: 3 <span style="white-space:pre-wrap"> </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 style="white-space:pre-wrap"> </span></div><div>missed calls indicator disabled: 1<span style="white-space:pre-wrap"> </span></div>
<div>suppress dtmf playback: 0<span style="white-space:pre-wrap"> </span></div><div>#audio mode: 2<span style="white-space:pre-wrap"> </span>#0 = speaker (default)1 = headset 2 = speaker/headset 3 = headset/speaker</div>
<div>time server disabled: 0<span style="white-space:pre-wrap"> </span></div><div>time server1: <a href="http://pool.ntp.org" target="_blank">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 style="white-space:pre-wrap"> </span></div>
<div>sip line1 proxy ip: <a href="http://aastra.mydomain.com" target="_blank">aastra.mydomain.com</a></div><div>#sip line1 proxy port: 5060</div><div>sip line1 registrar ip: <a href="http://aastra.mydomain.com" target="_blank">aastra.mydomain.com</a></div>
<div>sip line1 registration period: 300</div><div>#sip line1 registrar port: 5060</div></div></font></div><div><font 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" target="_blank">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" target="_blank">aastra.mydomain.com</a></div><div>MWI-Account: <a href="mailto:203@aastra.mydomain.com" target="_blank">203@aastra.mydomain.com</a></div>
<div><br></div><div>Call-ID: d8fd4c8795801cd2</div><div>User: <a href="mailto:203@aastra.mydomain.com" target="_blank">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" target="_blank">aastra.mydomain.com</a></div><div>MWI-Account: <a href="mailto:203@aastra.mydomain.com" target="_blank">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" target="_blank">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" target="_blank">aastra.mydomain.com</a></div><div>MWI-Account: <a href="mailto:203@aastra.mydomain.com" target="_blank">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><div></div><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" target="_blank">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><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>> ------------------------------------------------------------------------<br>
><br>
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