I have a new system that I am configuring. At the moment I have 5
Polycom 650s connected to a sipX 4.0 box, all on the same private
subnet. All sipX components including the conference server are on the
same box. I am connected to Voip.ms through sipXbridge.
I am testing a conference room a
On Fri, 2009-05-22 at 13:25 -0400, Hansen, Hans wrote:
> Still doesn't work.
>
> [r...@gmtsipx01 ciscoAta]# ls -l /etc/sipxpbx/ciscoAta/cfgfmt
> -rwxr-xr-x 1 sipxchange sipxchange 26068 Dec 15 2003
> /etc/sipxpbx/ciscoAta/cfgfmt
>
> Error still is
> java.io.IOException: Cannot run program "/etc
On Tue, 2009-05-19 at 21:13 -0700, Jason Jason wrote:
> I am using Sipxecs version 3.10.2-013143 2008-07-23T18:09:14
> ecs-centos5 with an Audicodes Mediant 600 Gateway 5.40A.020 for the
> SIP Gateway and Nortel 6830 IP Phone with firmware version "V1238".
>
> In several upscale systems, there is
On Thu, 2009-05-21 at 15:32 -0500, Goran Donev wrote:
> What am I missing.
You failed to follow this section of the instructions:
> Attach the snapshot(s) to the issue, along with the problem
> description; make sure you include:
>
> - What you did
> Include the identifying information
Hi I'm running SIPX on a virtual machine, and I'm in the
process of testing it to later deploy it in a small office environment.
My problem is that I'm unable to register xlite soft phones with the SIPX
server, I get the error "Registration error 404 not found". And this
is happening for any ot
Still doesn't work.
[r...@gmtsipx01 ciscoAta]# ls -l /etc/sipxpbx/ciscoAta/cfgfmt
-rwxr-xr-x 1 sipxchange sipxchange 26068 Dec 15 2003
/etc/sipxpbx/ciscoAta/cfgfmt
Error still is
java.io.IOException: Cannot run program "/etc/sipxpbx/ciscoAta/cfgfmt":
java.io.IOException: error=2, No such file
Hi folks,
I'm trying to understand HA in sipXecs 4. I had a working configuration
with 3.10.3 but now it appears
that my tcp wrappers is interfering with sipxcallresolver connecting to
the database.
On the Master:
sipxcallresolver.log
"2009-05-22T09:09:29.214654 ":DEBUG:Read CSEs from
rename it to something else then back to
cfgfmt (no dot or extension whatsoever).
What is the output of?
ls -l /etc/sipxpbx/ciscoAta/cfgfmt
>>> "Hansen, Hans" 05/22/09 11:54 AM >>>
Same error message.
Hans C. Hansen
Manager - IT Services
20420 Century Blvd
Germantown, MD 20874
Phone 301-9
On Fri, 2009-05-22 at 14:11 +0100, Damian Dowling wrote:
> However, I find storing the phone profiles only on the primary server
> surprising. It would improve redundancy if the profiles were also stored
> on the secondary server. Is this planned for the future?
As a general rule, if a phone is re
Same error message.
Hans C. Hansen
Manager - IT Services
20420 Century Blvd
Germantown, MD 20874
Phone 301-944-2790
hhan...@currentgroup.com
www.currentgroup.com
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Friday, May 22, 2009 11:48 AM
To: Hansen
Try this.
cd /etc/sipxpbx/ciscoAta
chown sipxchange:sipxchange cfgfmt
chmod 755 cfgfmt
>>> "Hansen, Hans" 05/22/09 11:39 AM >>>
Already did. Tried root and sipxchange as the owner/group. Full rights 777.
Hans C. Hansen
Manager - IT Services
20420 Century Blvd
Germantown, MD 20874
Phone 301-
Damian Krzeminski wrote:
> In addition to what Scott said: If you want it you can set up a
> provisioning server separate from a primary server even today (replicate
> provisioning directory, set-up FTP/TFTP/HTTP servers, set up DNS so that
> phones find the best provisioning server.
> I saw little
The wiki article has this included:
"What I also did was to extract the cfgfmt.linux file, and place itin
the /etc/sipxpbx/ciscoAta directory, giving it execute rights andmaking
it owned by user/group sipxchange, and renamed it to cfgfmt(remove the
�.linux�)."
http://sipx-wiki.calivia.com/index.p
Already did. Tried root and sipxchange as the owner/group. Full rights 777.
Hans C. Hansen
Manager - IT Services
20420 Century Blvd
Germantown, MD 20874
Phone 301-944-2790
hhan...@currentgroup.com
www.currentgroup.com
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepar
Change the name to cfgmgmt (no extension) and make sure it has execute rights.
>>> "Hansen, Hans" 05/22/09 11:33 AM >>>
Can't Configure Cisco ATA 188. I downloaded the cfgfmt.linux and put
that and the ata-ptag.dat file in the /etc/sipxpbx/ciscoAta folder.
The error I get is : java.io.IOExce
Can't Configure Cisco ATA 188. I downloaded the cfgfmt.linux and put
that and the ata-ptag.dat file in the /etc/sipxpbx/ciscoAta folder.
The error I get is : java.io.IOException: Cannot run program
"/etc/sipxpbx/ciscoAta/cfgfmt": java.io.IOException: error=2, No such
file or directory
***CON
I've done some additional testing, without doing any changes on the
Cisco side, I reloaded SipX and reconfigured it and I had the exact same
issue where I could not call from the Cisco CME to the SipX extensions.
Luckily, I took snapshots of every related page before I reloaded SipX
and was a
Scott Lawrence wrote:
> On Fri, 2009-05-22 at 14:11 +0100, Damian Dowling wrote:
>
>> SipXconfig running on a single server I can understand, it makes
>> everything a lot simpler from a programming point of view. Although it
>> would be nice for load balancing reasons to allow users to access thei
I have fount the Sipx relay to have little impact on call quality, other than
your typical QOS issues on a low bandwidth internet connection. But thats not
Sipx's fault.
The difference with Asterisk, is that Asterisk transcodes everything to IAX,
where as Sipx is just relaying the traffic.
So
On Fri, 2009-05-22 at 07:19 +0800, Ronny Tjoa wrote:
> Scott, thanks for your analysis on each siptrace file. As for the Contact *,
> I have filed that issue.
>
> Is something wrong with the newsgroup ? I cannot see my previous post with
> sipviewer xml attachments.
At least one made it to the
Scott Lawrence wrote:
> On Fri, 2009-05-22 at 14:11 +0100, Damian Dowling wrote:
>
>
>> SipXconfig running on a single server I can understand, it makes
>> everything a lot simpler from a programming point of view. Although it
>> would be nice for load balancing reasons to allow users to access
On Fri, 2009-05-22 at 14:11 +0100, Damian Dowling wrote:
> SipXconfig running on a single server I can understand, it makes
> everything a lot simpler from a programming point of view. Although it
> would be nice for load balancing reasons to allow users to access their
> accounts via different se
On Fri, 2009-05-22 at 23:36 +1200, Yakout Esmat wrote:
> Hi,
>
>
>
> I would like to configure a pickup group or even directed pickup group
> on sipX 4.0.
>
A feature we don't have yet - see
http://track.sipfoundry.org/browse/XX-4775
___
sipx-us
On Fri, 2009-05-22 at 23:32 +1200, Yakout Esmat wrote:
>
> If we had 2 endpoints separated by NAT, hence media has to travel
> through the Media Relay agent (the sipX server) would the call quality
> be better than if the same endpoints were using an Asterisk server?
The real answer is probably t
Damian Krzeminski wrote:
> Damian Dowling wrote:
>
>> Hi,
>>
>> I am currently setting up the HA system using SipXecs version 4 and was
>> wondering if someone could verify a couple of things.
>>
>> 1. You can only login to the Primary server via a web browser, the
>> Secondary server will refus
hazzanz wrote:
> Is it possible in 4.0 to move the voicemail server to the a secondary
> server?
>
> Thanks
Current stable 4.0 has a bug:
http://track.sipfoundry.org/browse/XX-5628
You need to wait for a stable 4.0.1 to do that.
D.
___
sipx-users ma
Damian Dowling wrote:
> Hi,
>
> I am currently setting up the HA system using SipXecs version 4 and was
> wondering if someone could verify a couple of things.
>
> 1. You can only login to the Primary server via a web browser, the
> Secondary server will refuse the connect on port 8443. Is this c
Hi,
I would like to configure a pickup group or even directed pickup group on
sipX 4.0.
All I could think of are the following:
1- Use the user group to bind users together where they can pickup
calls from each other's extensions. But that on its own is not enough, will
need to con
Hi All,
So far we have been avid users of Asterisk-based PBXs with all our clients
running distributions like Trixbox and PIAF.
But now with the release of the new sipX version 4.0, I believe that we have
to revisit our strategies.
I have many questions but let's focus on quality issue
Hi,
I am currently setting up the HA system using SipXecs version 4 and was
wondering if someone could verify a couple of things.
1. You can only login to the Primary server via a web browser, the
Secondary server will refuse the connect on port 8443. Is this correct?
2. Device Files and Phone P
Is it possible in 4.0 to move the voicemail server to the a secondary
server?
Thanks
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