[Freeswitch-users] Aastra phone registration lost
Aloysius Lloyd
lloyd.aloysius at gmail.com
Tue Feb 15 18:42:48 MSK 2011
I have the same issue, around 275 phones in the field. I want the 275 phones
work with FreeSWITCH.
Thanks
Lloyd
On Tue, Feb 15, 2011 at 10:29 AM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:
> Good luck!
>
> There are some aastra.cfg sections earlier in this thread. One thing I
> found about the 9133i and other legacy phones is that they work better
> if you don't subscribe to MWI but instead send unsolicited updates.
> They also have odd default codec settings, so you want to specify the
> codec string with the ptime that you want (I think they default to 30
> instead of 20).
>
> I like the newer phones much better, but we have about 150 of these in
> the field, so we have to make them work.
>
> -Tim
>
> Aloysius Lloyd wrote:
> > 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 <mailto:fs-list at communicatefreely.net>>
> > wrote:
> >
> > <domains>
> > <domain name="all" alias="true" parse="true"/>
> > <domain name="pbx.MYDOMAIN.NET <http://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>
> > <mailto: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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