[Freeswitch-users] Aastra phone registration lost

Tim St. Pierre fs-list at communicatefreely.net
Tue Feb 15 02:53:41 MSK 2011

    <domain name="all" alias="true" parse="true"/>
    <domain name="pbx.MYDOMAIN.NET" alias="true" parse="false"/>

    <param name="user-agent-string" value="Communicate Freely 2.0"/>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="log-auth-failures" value="true"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${internal_sip_port}"/>
    <param name="dialplan" value="xml"/>
    <param name="context" value="internal"/>
    <param name="dtmf-duration" value="100"/>
    <param name="inbound-codec-prefs" value="$${internal_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${internal_codec_prefs}"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="sip-ip" value="$${public_ip}"/>
    <param name="rtp-ip" value="$${public_ip}"/>
    <param name="hold-music" value="$${moh_prefix}alt"/>
    <param name="dtmf-type" value="rfc2833"/>

    <param name="force-register-domain" value="$${domain}"/>
    <param name="force-subscription-domain" value="pbx.$${domain}"/>
    <param name="force-register-db-domain" value="$${domain}"/>
    <param name="force-subscription-expires" value="600"/>

These are the most important ones I think.

    <param name="NDLB-received-in-nat-reg-contact" value="true"/>
    <param name="sip-force-contact" value=" NDLB-connectile-dysfunction"/>

I'm also using sip-force-expires to 600 at the moment, and ping = 10.  I 
will probably increase those eventually to reduce bandwidth.  I'm still 
in Beta right now, but I'm not having too many issues.

Some of those params are added per-device using the directory, so I can 
tweak them depending on which device registers, and what the NAT status 
is of that device.  I'm really pushing to get IPv6 on the phones, as 
well as on some of the more prominent (but competitive) DSL providers 
here so that we can forego NAT altogether some day.

Hope that's helpful.  I haven't really gone through and figured out 
which variables do what at the moment, but it seems to work as it is.

Aloysius Lloyd wrote:
> Tim,
> Thank you for the information.
> I have around 275 Aastra 9133i models phones in production .These 
> phones installed 31/2 years ago. I am trying to migrate to FreeSWITCH 
> could not make it work reliably.
> What are the profile settings turned on for these phones works reliably?
> Thanks 
> Lloyd
> On Fri, Feb 11, 2011 at 12:54 PM, Tim St. Pierre 
> <fs-list at communicatefreely.net <mailto: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
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