What if you were to route that DID (anonymous) to an Auto Attendant that
played a reorder tone and a message stating that anonymous callers are
not permitted?
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Austin
Curry
Sent:
On Fri, 2009-11-20 at 12:38 -0600, Austin Curry wrote:
Does anyone have a solution for blocking incoming calls that pass no
digits but caller ID shows up as anonymous or similar?
Not yet...
http://track.sipfoundry.org/browse/XX-5076
___
Hi, All
I have one question about the sipXecs stack:
Is this sip stack supporting IPv4/IPv6 dual-stack receiving spontaneously?
I mean whether the Sip messages from IPv4 and IPv6 hosts can be received at the
same
time？
Thanks very much!
Best regards
Charles Zhang
On Sat, 2009-11-21 at 20:18 +0800, c z wrote:
Hi, All
I have one question about the sipXecs stack:
Is this sip stack supporting IPv4/IPv6 dual-stack receiving
spontaneously?
I mean whether the Sip messages from IPv4 and IPv6 hosts can be
received at the same
time？
sipXecs does not
You will need to answer some things before you begin.
Does FLOWROUTE have the ability to send you calls on port 5080? Do they
require registration or do they send you calls by IP address instead? Does
your firewall in front of sipx have a static IP address?
You will also need to understand what
Tony,
Thanks for the reply!
I guess what concerns me is that the memory allocation, in my case, is
instantaneous. It doesn't take a day or two, more like a *second* or two. :)
My situation is that I am attempting to overcome the non-existence of
true multi-tenant support (not meant as an
On Fri, 2009-11-20 at 23:29 -0800, Jordan Turner wrote:
After re-reading the advices, I redid my configs differently. I
started with vanilla install again and this time I did not choose
Internet Calling. And the extension calling between remote workers
worked! somehow, I had in my mind that
Updating : sipxconfig-ftp[163/361]
error: %post(sipxconfig-ftp-4.0.4-017289.i386) scriptlet failed, exit status
1
The package is installed. Is this cosmetic or should I be concerned?
--
==
Tony
Guys I have tried and tried to find a reasonable solution to a PRI media
gateway in the OS realm Asterisk Freeswitch ect but still can not find a
solution to forwards to mobile devices via a users forwarding options as well
as unpredictable assisted transfers
Note there is no nat in this
Stop and ensure you are NOT using firmware later than 3.1.3Rev3. You say
latest, and 3.2 is known to have issues which polycom if working on.
IMO hardware gateways (I prefer patton over audiocodes), because they. Are
much less dependent on moving parts, and are more reliable.
I didn't know AC
On Sat, Nov 21, 2009 at 11:37 AM, Gabe Casey
gca...@franklinamerican.com wrote:
Guys I have tried and tried to find a reasonable solution to a PRI media
gateway in the OS realm Asterisk Freeswitch ect but still can not find a
solution to forwards to mobile devices via a users forwarding options
Tony,
Okay -- working theory here. The thing in my environment which is
different is that I am using OpenVZ.
I started running into issues with *fresh* installs immediately failing
on service start-up. All kinds of Java JVM memory allocation errors.
First of all, with Sun's java is
When you start having swap space you need to understand the difference
between available and in use.
I've always understood that when swap becomes in use, you would begin to
experience performance degradation related to the percentage in use.
Hope that your changes work out for you, looks
Tony,
My point is that OpenVZ containers will not show *any* swap space
available or in use, because there isn't any. There can't be because
containers don't work like that. If this were a KVM or VMware virtual
machine (something powered by a hypervisor of sorts), obviously there's
be a swap
Gabe,
This looks to be experimental but how about this:
http://sipx-wiki.calivia.com/index.php/How_To_Install_Sangoma_WANPIPE
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gabe Casey
Sent: Saturday, November 21,
Yes it is very possibly the issue at hand that is being manifested in
other situations as well.
cc : sipx-users as this is relevant to your original posting.
On Sat, Nov 21, 2009 at 12:24 PM, Gabe Casey
gca...@franklinamerican.com wrote:
The Polycoms are running Version 3.2.1.0054. I look at
Yes, I do register with ITSP just fine.
Attached you will find Flowroute's config per your request. Thank you.
--- On Sat, 11/21/09, M. Ranganathan mra...@gmail.com wrote:
From: M. Ranganathan mra...@gmail.com
Subject: Re: [sipx-users] sipX Bridge
To: Jordan Turner
A while back, someone had posted asking if there were any integrations for sipx
but I
don't recall seeing any or many replies. Are there?
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive:
Matt,
Yeah -- I'd never use less than 1GB.
But here's the thing that I feel might have given you the wrong idea
about what I am doing:
HostContainer
Physical256 2048
Swap17920
OpenVZ does not give the container any swap space. It can't do that
because it's
Here are the three advantages I've seen to using the new FreeSWITCH
voicemail system over the old one so far:
Better Performance
IMAP integration
HD Audio
Are there other functional advantages to the new voicemail system? Will
it have shiny new things no one has talked about yet?
What
Folks,
I'm impressed with the array of support for Polycom handsets. That's
what I've been wanting to base my deployments off of because of the HD
voice support.
I know that Aastra now supports G.722 wideband. But Aastra's entry-level
handsets have a big deal-breaker for me -- those dang
I agree... They don't look half bad. I still see some paper inserts, but
they are not garish like the Aastra ones.
I'm told they are very inexpensive and reasonably well built. The fact
that even the basic ones support G.722 is important. That's a killer
feature to demonstrate to clients.
FYI...the aastra does not support MoH or BLF.
Other than that they work ok. I've hit several bugs with their phones but they
where not SipX specific.
-M
On 11/21/2009 at 06:07 PM, in message 4b0872b9.9070...@spudland.com,
Robert B d...@spudland.com wrote:
I agree... They don't look
Wow it looks like Cisco and Alcatel had a litter.
As long as it properly supports PRESENCE and eventlist/broadsoft BLF
then BLF should work just fine. I don't see any information about what
types of BLF it supports. The only thing you can really do is buy two
of them and see if all the
24 matches
Mail list logo
