Any complaints about this one?
http://www.voiplink.com/Audiocodes_MP_118_FXO_p/audiocodes-mp-118-fxo.htm
Tony Graziano wrote:
> I prefer patton for this. Its nice to be able to get factory direct support.
>
> Tony Graziano, Manager
> Telephone: 434.984.8430
> Fax: 434.
On Fri, 11 Dec 2009 15:01:00 -0800, Dave Deutschman wrote:
> I take it this is an AudioCodes Mediant 2000? I checked with my AudioCodes
It's actually a 5000 but I'm told the cards function in either the 2000 or 5000
and even others.
Still researching that.
> I take I you bought yours some
> ti
Thanks. Is this the right product:
http://www.patton.com/products/pe_products.asp?category=51&MiDAS_SessionID=bdd176a997e24043a446e3938f89d84d
--Original Message--
From: Tony Graziano
To: mkitchin.pub...@gmail.com
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] ATA device for e
I prefer patton for this. Its nice to be able to get factory direct support.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.
I know plenty of you have done this, so I just wanted to know what hardware
people have had the best experiences with. We are on sip only from the carrier
at the moment. We want to put 4 to 6 backup lines in. They are being delivered
on analog lines (split of a pri) using this:
http://www.adtran
I'd suggest you start with your network card. Seeing your hardware choice
makes me ask if this is running bare metal or virtual. If bare metal there
is likely a driver or mtu issuewith the nic, or a cable/port issue.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 4
On Fri, 2009-12-11 at 17:27 -0500, Geoff Van Brunt wrote:
> We have been having an issue where voice mail messages get cut off. On
> the callers side, the start recording their message and then at some
> point they get transferred and it starts ringing, with no answer. The
> length of time before t
Well, the phone (a Polycom 450) is getting an IP address so it's able to
talk to the DHCP server successfully.
I don't know why you brought up DNS. DNS is working OK.
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.si
The "Option 66" and "Option 120" are separate tests. There's different
preflight options to run them.
Interestingly, these tests succeeded a few days ago, before I decided to
re-install sipXecs from the CD and try again.
The machine that I'm running sipXecs on seems to have proper network
connect
Updating to 4.0.4 is certainly not going to hurt but...
What kind of phones and what type of transfer is occuring? Be sure to
include bootrom and firmware versions.
It sounds like a transfer sequence is not completing. It could be the phone
OR the user.
Tony Grazi
We have been having an issue where voice mail messages get cut off. On
the callers side, the start recording their message and then at some
point they get transferred and it starts ringing, with no answer. The
length of time before the transfer seems random. I have yet to be able
to capture a trace
On Fri, 2009-12-11 at 14:20 -0800, Robert Swirsky-Warner wrote:
> Any ideas?
Take a network trace to see what DHCP traffic the test is putting on the
wire.
Dale
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipf
its likely DNS will faul on end user devices too, seems you skipped a big
fat README.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax:
Its looking for options that aren't configured on your dhcp server in your
lan.
Bootp
Option 66 and 120 also.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Tel
The DHCP pre-flight test fails on the server (a CentOS Linux box running the
CentOS + sipXecs distro.)
[r...@sipx ~]# /usr/bin/preflight --dhcp-test -v
Starting DHCP server test.
Sending DHCPDISCOVER request.
Read timeout: Receive timed out
Network timeout. Retrying with timeout increased to 2
Folks:
I've been unable to get sipXecs to do anything. Is there anyone in the
Sunnyvale, CA area who does this for a living and can spend a few hours
getting this to work for me?
This is a VERY simple configuration: 1 server, (running the CentOS
single-disk install on a dedicated MacMini) and 1 P
We have been testing SIPX in a lab for some time and it has been performing
very very well as a possibility for a replacement of our current PBX. However,
one piece that Cisco CM does well is showing the status of each users phone.
Cisco CM has a small app that runs on the desktop and each user
Just thinking out loud.
I have a customer that is using an audiocodes device. They also have a lot
older cisco phones that do not support the MoH URI setting. The audiocodes
does not support this setting either.
They really want MoH.
So I'm wondering if I could simply create a sipxbridge
Anyone know if sipxecs works just fine or not with mediant version 4.6?
I have one unit which is running 5.6 which seems to be working fine but I have
another which is only 4.6 and am told would need a memory upgrade if I wanted
to run 5.4 at least. Not sure what that involves so wondering if 4.
On Fri, 2009-12-11 at 15:13 -0500, Lara Johnson wrote:
> The hard phones in the office had their profiles reconfigured by the
> sipx box. I had to resend the profiles to get them to register.
> Polycom 500's.
>
> The softphones have the domain name configured not the FQDN. So
> everything should
The hard phones in the office had their profiles reconfigured by the sipx box.
I had to resend the profiles to get them to register. Polycom 500's.
The softphones have the domain name configured not the FQDN. So everything
should be just domain.biz. The new box was set up with domain.biz as the
On Fri, 2009-12-11 at 14:47 -0500, Lara Johnson wrote:
> That's what I thought initially too, so I set up a brand new box and actually
> took the phone system down and switched and made sure I didn't start out with
> the fully qualified domain name. It still has the same problem.
>
> Setting it
That's what I thought initially too, so I set up a brand new box and actually
took the phone system down and switched and made sure I didn't start out with
the fully qualified domain name. It still has the same problem.
Setting it up initially for the domain name should have set the realm to th
Will be in 4.2
http://track.sipfoundry.org/browse/XX-5594
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jordan
Turner
Sent: Friday, December 11, 2009 12:12 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx
You may have to put a static route in the PBX to route back out through
that firewall to go to the ITSP's IP addresses.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jeff
Gilmore
Sent: Thursday, December 10, 200
On Fri, 2009-12-11 at 13:26 -0500, Lara Johnson wrote:
> Hopefully this helps. I don't know what else to try to look at.
I think your problem is that you've gotten your 'realm' out of sync.
Everyone has to be using the same value - all your servers, all your
phones both inside and outside.
The
Hopefully this helps. I don't know what else to try to look at. The trace
didn't show up much. Hope someone can help me out. Thanks.
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: Friday, December 11, 2009 11:22 AM
To: Lara Johnson
Cc: Tony Graziano; sip
On Thu, 2009-12-10 at 16:52 -0500, Burden, Mike wrote:
> I have a Trunk Hunt group set up at extension 99 that rings all of our
> “hard” phones simultaneously.
>
> The destination for inbound calls is set up as 99 in the sipXbridge.
>
>
>
> The CDR report shows all inbound calls as having “fai
Other than the CDR, the system works exactly as expected.
The CDR shows a failed call to extension 99, and no activity to any
other extension at that time (including the one that answered the call.)
Not an issue that I really need to have solved... just poking around.
:)
Mike Burden
Lynk Syste
sorry, cancel inquiry about Conference Recordingfound info.
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-u
Is recording of conference calls possible?
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
sipXecs IP PBX -
On Fri, 2009-12-11 at 11:10 -0500, Lara Johnson wrote:
>
> I currently have all the permissions on all of my dial plans set to
> none. Which solved the outbound problem with the external users in
> 3.10.3. Where a call would look like it was coming from and going to
> the same place (the ingate).
Then your external users are registering with an alias?
We use the same setup internally here with out ingate. Never an issue with
that. BUT, our remote users (polycom and/or xlite, bria pro) never register
or have a contact address or an alias. The contact and the registration
should be the user
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Friday, December 11, 2009 9:50 AM
To: Scott Lawrence
Cc: Lara Johnson; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 4.0.2 Remote users & Authentication Realm problem
On Fri, Dec 11, 2009 at 9:46 AM, Scott Lawrence
mailt
On Fri, Dec 11, 2009 at 9:46 AM, Scott Lawrence
wrote:
> On Fri, 2009-12-11 at 09:05 -0500, Lara Johnson wrote:
> > Well, I tested the theory. I set up a brand new box and configured it
> > exactly how my production system ran when I did. External users that
> > are coming in through my ingate can
On Fri, 2009-12-11 at 09:05 -0500, Lara Johnson wrote:
> Well, I tested the theory. I set up a brand new box and configured it
> exactly how my production system ran when I did. External users that
> are coming in through my ingate can register (you can see them in the
> register). And they can rec
You should be OK as long as you know enough about networking to send
traffic destined for the ITSP network out of that dedicated connection
or if your router/firewall supports 1 to 1 NAT and has a secondary
interface for your ITSP connection, utilize that instead. All you really
need to do afte
Well, I tested the theory. I set up a brand new box and configured it exactly
how my production system ran when I did. External users that are coming in
through my ingate can register (you can see them in the register). And they can
receive calls from internal users.
Though when they try to cal
38 matches
Mail list logo
