[Freeswitch-users] Aastra phone registration lost

Tue Feb 15 06:50:20 MSK 2011

Tim,

Thank you for the settings will give a try.

Thanks
Lloyd

On Mon, Feb 14, 2011 at 6:53 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

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