[Freeswitch-users] Aastra phone registration lost
Aloysius Lloyd
lloyd.aloysius at gmail.com
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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> >
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