Nitin Mirchandani wrote:
On a user list - I am looking for onstructive ideas as I feel the
project I have is quite complex(atleast for me) and I will be ditching
trix for sure as its a admin hell.
Sipx seems clean and simple but as I have nil experience, need "step
by step" help.
Ok well here is the conclusion we've reached so far...
You should set up a normal SipXecs system (using version 4.0.1) at Site
A, and add all of your users from all of your sites to it. Then add a
secondary server, and put it at Site B. The two sites must be linked
with VPN. This primary/secondary arrangement effectively means that the
two servers will act as one, so there's no need for site-to-site dial
plans, or multiple lines on each phone. This arrangement is also easily
expandable, as you can add additional secondary servers for each site
you wish to add. I believe that the phones at Site A will need to be
given the registrar IP address of the primary server, and the phones at
Site B will need to be given the registrar IP address of the secondary
server. Scott will be able to confirm that.
Out of the phones that you mentioned, The Polycom SoundPoint 550 is
definitely supported as a "managed" phone in SipXecs. Not sure about
the Cisco 7950, but even if its not supported as a "managed" phone, it
will be supported as an "unmanaged" phone. I recommend that you add all
of the managed phones to your SipXecs server (via the MAC address of the
phone), and use SipXecs to centrally manage their configuration via
auto-provisioning. If you do this, you will need to make a few
adjustments to your DHCP server (to tell it the provisioning URL). If
SipXecs is acting as your DHCP server then this will all be done
automatically, but if not then you'll have to configure it manually. It
is possible to get auto-provisioning to work without making the
adjustments to the DHCP server, but you would need to manually configure
the phones as a one-off, in order to put the provisioning URL into them.
You can set up your dial plans however you require them for accessing
the FXO gateways, but when you do so, bare in mind that the same dial
plan will be applied to all users at all sites. You can create
permissions and apply them to particular user groups if necessary.
I'm certain that what I've described above is the best solution for you,
but there are some limitations you need to be aware of:
1. You will not be able to use a different codec through the VPN link
than you do between phones at the same site. Whatever you choose
as your main codec will be used regardless.
2. Media services such as voicemail and auto-attendants are only
handled by the primary server. If a call to them originates from
one of the other sites, that call will be routed through the VPN
link back to the primary server.
3. The system is only configurable via the web interface on the
primary server. Secondary servers don't have web interfaces.
I hope that points you in the right direction at least. You will find a
lot of the more specific implementation information on the wiki at
http://sipx-wiki.calivia.com/
Best regards,
Keith.
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