Does anybody have experience with the Audiocodes Mediant 600 gateway (with
PRI) and SipX?
Any feedback would be greatly appreciated.
Cheers,
Marcello
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0 Oct 2010, Marcello Manzardo wrote:
> Anybody any great ideas on how to handle 411 and 555-1212 calls
> besides from just blocking???
It there really a need for '411' when prevasive access to search engines
exists?
We did not block prohibited outbound numbers (out of permited area
: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Marcello
Manzardo
Sent: Tuesday, July 27, 2010 8:25 AM
To: 'Josh Patten'; sipx-users@list.sipfoundry.org
Cc: b...@mmdbiz.com
Subject: Re: [sipx-users] SipX and Xen Server Tools
Thank you
://download.ezuce.com/sipfoundry/4.2.1/sipxecs-centos.repo and put it in
/etc/yum.repos.d) then run 'yum install sipxecs'
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/27/2010 12:11 AM, Marcello Manzardo wrote:
I have been running SipX on
I have been running SipX on Xen Server successfully for quite some time.
However, there is one nagging issue that prevents LiveMotion which requires
the Xen Tools to be installed.
Did anybody figured out a way to install Xen Tools on a SipX virtual
machine???
Any help is greatly appreciate
Thanks for all the detailed instructions.
Just wanted to let everybody know about the issues I encountered and how I
was able to work around it.
. This was a clean 4.2.0-018575 install from ISO
. Changing the sipxecs.repo file to include the new base url for
[sipxecs-stable] res
Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: Marcello Manzardo
To: 'Tony Graziano'
Cc: 'Stephen D. Miller' ; 'D. R. Lang'
; sipx-users@list.sipfoundry.org
Sent: Sat May 15 12:46:50 2010
Subject: RE: [sipx-users] Java goes to 89 p
perly showing me details and
advanced settings, and getting slow, unless I had done so.
On Sat, May 15, 2010 at 12:26 PM, Marcello Manzardo
wrote:
> I have seem the same issue numerous times. Most of the time I can cure it
by
> restarting the management service and of course a complete system rest
I have seem the same issue numerous times. Most of the time I can cure it by
restarting the management service and of course a complete system restart
fixes it.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ste
Hi Mike,
I tested the HT503 FXS and FXO outbound only.
FXS: Works fine no apparent issues detected
FXO: I only was planning on using it as E911 emergency outbound trunk which
works "fine".
- It was setup as an unmanaged GW
- Hang-up is not detected if the remote site picks up; for example if I
Just in case here is the link:
http://sipx-wiki.calivia.com/index.php/SipX_ConfigServer_Customize_Colors,_L
ayout_and_Logo
Marcello
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Monday, May 0
With Audacity, (opensource sound editor, http://audacity.sourceforge.net/ )
you should be able to adjust the volume without any problems.
Drag and drop the file into audacity, make sure to set the output parameters
to
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bits, mono, 8000 Hz
nd
the PIN as the password. this will get you to the end-user portal.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Marcello
Manzardo
Sent: Friday, April 30, 2010 8:04 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users]
These may be really dumb question but I was not able to find answers.
1. Under User Group Settings, Conference, you can automatically assign
conferences to new users.
I tested this, added a new user but could not find a "conference extension"
anywhere; so I must be missing a step???
2.
osh Patten
> wrote:
>
>
> Also, I have a Polycom 330 that I use exclusively for
testing and it does
> everything it should. The 330 is an older version of the 331, which is
> essentially a 321 with an extra ethernet port for connecting the computer
> to. I have
on hold does
either)
If you're planning on buying stick with Polycom, end of story.
Marcello Manzardo wrote:
Does anybody have experience with Aastra 6730i phones and sipX 4.2?
Any feedback would be greatly appreciated.
Does anybody have experience with Aastra 6730i phones and sipX 4.2?
Any feedback would be greatly appreciated.
Thanks,
Marcello
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I used the suggestions as well and they do work nicely :-)
For the resulting call I use 18004664411 (the free Google service, as
suggested earlier in the thread) which is better than a failed call.
Also don't forget to make a similar rule for 411...
Cheers, Marcello
-Original Message-
Fr
I cannot get MWI to work on any Siemens optiPoint phones (400, 600 or 420).
In the phone configuration there is field "Message Waiting IP address: "
which is set to 0.0.0.0
This is the setting that used to work with an Asterisk system.
I tried to enter the sipX IP address in that field but then
-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Wednesday, March 10, 2010 7:04 PM
To: marce...@discsox.com
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Failing registeration with Telasip ITSP
On Wed, Mar 10, 2010 at 5:15 PM, Marcello Manzardo
wrote:
> I am having trou
I am having trouble registering with Telasip ITSP.
Below is what their Tech Support told me:
. sipX seems to use the P-Asserted-Identity header for
authentication, which we cannot support.
. The userid must be in the from header, and callerid must be in the
Remote-Party-ID he
sipX.
Marcello
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Marcello Manzardo
Sent: Thursday, March 04, 2010 1:53 PM
To: 'Scott Lawrence'; 'M. Ranganathan'
Cc: sipx-users@list.sipfoundry.or
Cc: marce...@discsox.com; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Delayed call setup with Flow Route - Wrong field in
INVITE
On Thu, 2010-03-04 at 16:09 -0500, M. Ranganathan wrote:
> On Thu, Mar 4, 2010 at 3:57 PM, Marcello Manzardo
> wrote:
> > I am using 4.0.4-
I am using 4.0.4-017289 ecs-centos5
I have setup flowroute ITSP but experiencing about a 1 min delay until a
call goes through.
Tech support from flowroute pointed to a wrong field in the SIP INVITE that
is generated by sipX.
The field in question is the "Route" field that is not supposed
Has anybody managed to register a Siemens optiPoint 400 phone on sipx.
The optiPoint 400 registers fine on an Asterisk systems but not on sipx.
Any configuration hints would be greatly appreciated.
Cheers,
Marcello
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