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