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On Aug 28, 2012 8:38 PM, "Ken Ridley"
mailto:k...@federico.net>> wrote:
Ok, The system is updated, and blind transfers are working, but I noticed, that
th
k about our Internet Fax services!
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On Aug 13, 2012 3:43 PM, "Ken Ridley"
mailto:k...@federico.net>> wrote:
FYI: here is the exact Sipx Version sipXecs (4.4.0- 2011-08-14EDT19:47:12
do
] Blind Transfer gos to voicemail
iso isn't updated all the time... after you install from iso you need to 'yum
update'... always...
mike
On Tue, Aug 14, 2012 at 12:43 PM, Ken Ridley
mailto:k...@federico.net>> wrote:
I checked for a new version, but according to the site thi
net>
Fax: 434.465.6833
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On Aug 13, 2012 3:43 PM, "Ken R
FYI: here is the exact Sipx Version sipXecs (4.4.0- 2011-08-14EDT19:47:12
domU-12-31-39-04-D4-A1)
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ken Ridley
Sent: Monday, August 13, 2012 11:56 AM
To: sipx-users@list.sipfoundry.org
d
of ringing the cell phone, the caller is sent to system voicemail
Attended transfers work fine
Is there a setting that I can modify to fix this
Thanks,
Ken Ridley
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wrong.
You have 4GB of RAM, which is usually enough for a system with ~25 users (IMO).
So were you using swap? Did you restart PROXY or REGISTRAR before pushing the
shiny red button?
On Wed, May 9, 2012 at 11:51 AM, Ken Ridley wrote:
> We have locked down all access form the outside, and still th
are considerable larger today it might be your were the subject a a DOS
attack, inspeacting the larger logs would show signs of that.
On Thu, Apr 19, 2012 at 1:26 PM, Ken Ridley
mailto:k...@federico.net>> wrote:
I posted a while back about all of the phones deregistering at the same time
I posted a while back about all of the phones deregistering at the same time,
and was told that it was because I only had 1 GB of ram in my server
A week ago, I upgraded the hardware to this:
HP ProLiant ML110 G7 664723-S01 4U Micro Tower Entry-level Server - 1 x Core i3
i3-2120 3.3GHz - 4 GB RAM
for you. See how the whole
permgenspace things works without enough ram...
Good luck.
On Feb 25, 2012 8:02 PM, "Ken Ridley"
mailto:k...@federico.net>> wrote:
Douglas Hubler mailto:dhub...@ezuce.com>> "I assert, and I
could be proven wrong, that if you want best perfor
.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Saturday, February 18, 2012 4:26 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] All phone registrations expiring
You don't.
On Feb 18, 2012 6:56 PM, &q
software
Subject: Re: [sipx-users] All phone registrations expiring
You should not be using any swap!
On Feb 11, 2012 5:29 PM, "Ken Ridley"
mailto:k...@federico.net>> wrote:
This is the info I collected
Shouldn't it be using more of the swap?
top - 14:02:47 up 3 days, 23:01,
;
registration attempts (hint: look at the log sizes for the previous several
days, if there is a larger file, then LOOK at that file). If these are FAKE
registrations (script or dos attack), address this at your firewall.
On Fri, Feb 10, 2012 at 4:23 PM, Ken Ridley
mailto:k...@federico.net>>
you,
Ken Ridley
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sipx is) to see if
the packet for the dial attempt is even reaching sipx. ensure your phone and
public dns records are configured correctly.
On Sat, Jan 21, 2012 at 11:27 AM, Ken Ridley wrote:
> Some more info
>
> When we call the remote phone from an office extension 4 out of 5
>
:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ken Ridley
Sent: Friday, January 20, 2012 12:21 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Remote phone reregisters every 5 min
The phones in the office register 50 min, and the remote phone reregisters
every 5 min
I poked around,
The phones in the office register 50 min, and the remote phone reregisters
every 5 min
I poked around, but I was unable to find a way to increase this
Can this be changed?
Ken
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ket, bigger than maximum of 65535)
>
> What do I need to do
>
> -Original Message-
> From:
> sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org>
> [mailto:sipx-users-
> boun...@list.sipfoundry.org<mailto:boun...@list.sipfound
27;t know of anyone using sipx with their fusion platform as of yet due to
issues.
On Tue, Jan 10, 2012 at 4:05 PM, Ken Ridley wrote:
> ITSP is Broadvox
> The system has been connected to them with no problems, for about two
> months I captured packets behind the router to the sipx, an
to be damaged or corrupt.
(pcap: File has 100705613-byte packet, bigger than maximum of 65535)
What do I need to do
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ken Ridley
Sent: Tuesday, January 10, 2012 1
t 3:19 PM, Michael Picher wrote:
> well, you already tried restarting it...
>
> i'd do a capture on the box itself with tcpdump and then inspect the
> .pcap file and see if you can see/hear the dtmf.
>
> mike
>
>
> On Tue, Jan 10, 2012 at 3:06 PM, Ken Ridley wrote:
...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Tuesday, January 10, 2012 11:32 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] no DTMF
so it's not the server, it's either the gateway or the provider.
On Tue, Jan 10, 2012 at 2:27 PM, Ken Ridley
mailto:k...@fe
Sent: Tuesday, January 10, 2012 11:19 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] no DTMF
does it work from inside if you dial the AA extension?
On Tue, Jan 10, 2012 at 2:14 PM, Ken Ridley
mailto:k...@federico.net>> wrote:
This morning none of the options on the
you for your help,
Ken Ridley
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...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gerald Drouillard
Sent: Friday, December 30, 2011 3:37 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Sip Hacking?
On 12/30/2011 5:17 PM, Ken Ridley wrote:
I found about 30 failed calls
-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, December 30, 2011 2:25 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Sip Hacking?
I thought your options were covered when you asked earlier this week.
On Dec 30, 2011 5:17 PM, "Ken R
this the sipx way of blocking
invalid users, or did I just get luck this time
If I'm just lucky what can I do to prevent this?
Ken Ridley
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ade) and i don't know if it will work in a remote scenario.
I would suggest moving to more sip compliant phones though.
On Tue, Dec 27, 2011 at 9:55 PM, Ken Ridley
mailto:k...@federico.net>> wrote:
I have a Polycom 501 phone connected at a remote location, with no VPN - There
is no pr
?
Ken Ridley
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ist.sipfoundry.org]>
On Behalf Of Ken Ridley
Sent: Sunday, December 18, 2011 4:38 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Receiving CNAM
The ITSP is Broadvox, they are pretty flexible, and I am told they use a
national database, that is updated regula
all have the best CNAM databases even if they send it.
On Sun, Dec 18, 2011 at 3:17 PM, Ken Ridley
mailto:k...@federico.net>> wrote:
I believe the answer to this question is yes, but my boss wants me to verify
it, before we purchase the service
I have searched the Internet, the wiki, an
SP501 phone, from an ITSP
Ken Ridley
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to:tgrazi...@myitdepartment.net>> wrote:
Wow. Who is the itsp? I've never seen one that reads the sip header wrong like
that.
On Dec 13, 2011 8:13 PM, "Ken Ridley"
mailto:k...@federico.net>> wrote:
I just got off the phone with ITSp
They can populate the CNAM, but, that wi
t the CNAM to the COMPANY NAME on the account.
On Tue, Dec 13, 2011 at 7:26 PM, Ken Ridley
mailto:k...@federico.net>> wrote:
I have a Sipx system set up with a sip trunk using the internal sipxbridge
I have Gateway: Broadvox /SIP trunk /Default Caller ID set to 5551231234
When a user maks
last name, to the
company name, not sure what problems this will cause
Thanks,
Ken Ridley
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comes in, it displays the
callerID, number dialed, and the directory lookup for the greeting.
From:
sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org>
[mailto:sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.
We have 2 main numbers that need to be answered by a user with a different
greeting
Is there a way to let the user know what number the caller dialed before they
answer the call
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