Thanks! I guess when I looked at that I wasn't thinking of it applying to the
phone groups, etc. I was actually expecting it to be specific to each
phone/phone group.
Thank you for the clarification.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] O
in the phone? sure, check the phonebook menu item... note the phonebook
members / consumers.
On Thu, Sep 20, 2012 at 11:55 AM, Geoff Musgrave <
geoff.musgr...@cacionline.net> wrote:
> Is there any way to “hide” specific users from being automatically
> listed in the directory of the phone asid
Tony,
Yikes, seems like you've had a rough go. Thanks for the tips. I'll test
through those over the coming days and report back.
- Jeff
On Thu, Sep 20, 2012 at 5:03 PM, Tony Graziano wrote:
> Some might argue it "should" work this way too (as in how it works now),
> since the first user
Some might argue it "should" work this way too (as in how it works now),
since the first user does not have the security to send the call where the
second user does...
You might be sure to test attended transfers and directed call pickups.
That's where they failed before. They have gotten better
A typical and standard approach is probably OK for you,
1. Make a backup
2. Shutdown sipx services
3. yum update
4. reboot (if kernel is updated, which it will probably do)
5. After logging into sipxconfig send the server its profiles.
On Thu, Sep 20, 2012 at 2:53 PM, Rick Heinlein wrote:
> Hell
Hello SipX list,
I have a straight forward install of SipX (version 4.4.0-
2012-02-08EST09:10:08 ip-10-72-10-163). I am curious if I can issue a *yum
update* on my servers without any ill configuration settings or
compatibility affects? I have three servers, one primary and two secondary
serve
Is there any way to "hide" specific users from being automatically listed in
the directory of the phone aside from completely removing the directory?
Thanks for your help.
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://l
Michael,
I have a number of TA900-series Adtrans on my network, and I'm familiar
with most of the ins and outs of their configuration. You're right,
they're not perfect, but I've become quite satisfied with them. I'm
curious what issues you've encountered.
In this particular sipX test scenario,
I've found that using DND on poly com phones results in calls not going to
voicemail unless this is left unchecked. I think VM should pickup but it
goes fast busy instead. If unchecked this rings silently at phone.
Anyone else think this is broken?
--
~~
Tony Graziano, Manager
Te
Well, those adtrans are known to have some SIP issues in working with our
platform...
Josh can speak up but we've tested them every so often and from what I know
from about 4 months ago they still hadn't ironed out all of the signalling
issues they have.
Mike
On Thu, Sep 20, 2012 at 9:06 AM, Tod
That feature works today, as AFAIK, has worked for years. Incoming calls
to an extension, can have forwarding set to ring at the same time, or at the
end of a timed period. There is no magic to that, the feature just works.
Something in your particular setup is keeping that from working.
Is there a workaround to allow a user to forward externally sourced calls
to external numbers? Perhaps by disabling a permission requirement somehow?
- Jeff
On Wed, Sep 19, 2012 at 11:53 PM, Joegen Baclor wrote:
> This is a long standing issue and is all about 302 redirects not able to
> gr
Hello,
You can download pre-built version of the plugin at
http://siplabs.ru/downloads/sipXyealink.tar.gz to try it.
We are now working on re-designed version with different menu structure
(corresponding with phone menu not config file structure) which will come
with build scripts.
2012/9/20 Doug
13 matches
Mail list logo
