[Freeswitch-users] Aastra phone registration lost
Philippe Sultan
philippe.sultan at gmail.com
Mon Feb 14 01:00:40 MSK 2011
Tim,
Based on your input, I kept those settings :
sip registration period: 3600
sip registration renewal timer: 3000
The phone registers to FS every 10 minutes. However, any other
registration period does not work like it should. It's obvious to me
that the phone ignores the expires parameter given in the Contact HF
and actually uses what I believe to be its default registation period
(3600 sec).
I tried to insert an Expires HF to the 200 REGISTER response sent back
from FS, and then the phone re-registers based on the time given in
this Header Field. It therefore seems to me that the phone is buggy,
even though adjusting the two parameters like you suggested it
perfectly solves the registration issue.
Thanks again,
Philippe
On Fri, Feb 11, 2011 at 6:54 PM, Tim St. Pierre
<fs-list at communicatefreely.net> wrote:
> Sure,
>
> I normally administer about 300 Aastra phones, with every model they
> make represented.
>
> I have 22 connected to our Freeswitch "beta" system, which will
> eventually become production.
>
> All the endpoints are behind NAT without exception. There are a number
> of legacy 9133i and 480i phones on the network that don't have the newer
> NAT traversal features available, but this doesn't seem to be a
> problem. I have some of the nat traversal options turned on in the
> sofia profile though, so fs will send media back to the originating
> address and port.
>
> They have been quite reliable, and the sound quality has been excellent,
> with the newer phones using g722 at 16KHz.
>
> There are a few advanced features that I haven't had a chance to play
> with yet, but here's what I have working:
>
> Regular calls, in and out.
> Intercom calls (auto-answer to speaker phone)
> Automatic update of destination name and number (updates when checking
> voice mail, and when calling an extension). Only on newer phones
> Blind and attended transfer
> Music on hold
> SIP using udp or tcp (haven't tried TLS yet)
> Fewer issues with DTMF than with asterisk, using rfc2833 dtmf (no issues
> as of yet).
> BLF lamps work correctly, flashing when the phone rings, lit steady when
> they are on the phone.
> Distinctive ringing works.
> I haven't tried SLA yet, but Aastra recently released a firmware update
> that fixes a missing header, reported to have broken correct SLA
> operation. I'm hoping to test that in the next week or two.
>
> The phones provision very nicely - we auto generate config using PHP
> scripts that generate a config file on the fly from the user database.
> These are very easy phones to deploy in large installations, or to the
> outside world (not readily accessible). They have just added some new
> features that allow for remote diagnostics of the phones as well.
>
> There is a great deal of XML programmability in the phones too, which
> I'm starting to use for call control and other useful things (updating
> forwarding rules in the database, or conference and recording control
> using ESL).
>
> Hope that helps!
>
> -Tim
>
> Aloysius Lloyd wrote:
>> Tim,
>>
>> Can you share your success stories FreeSWITCH and Aastra.
>>
>> Aastra Phones Behind the NAT?
>>
>> In my case Aastra phones registration not a problem.
>>
>> But calls drooped every 60 sec ... in the same environment Linksys and
>> Polycom works perfectly.
>>
>> How stable the Aastra phones with FreeSWITCH system.
>>
>> TIA
>>
>> Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
--
Philippe Sultan
More information about the FreeSWITCH-users
mailing list