Hi Tony,
Thank you for your reply.
This is what I see in the SipXeconfig:Conference > New Conference
Name:
Extension:
Description:
Conference Owner
Participant PIN:
Maximum legs:
Music On Hold source:
Could you please explain what are the correct values?
Thanks in advance.
Rhon
On Sun, Apr 1
I've switched to KVM-over-IP (I use AdderLink or built in ipmi2/Raritan cards),
but you could go "old school" and just switch to using a serial console. You
only need to change a couple of places in linux to make the main system console
serial:
1) in /boot/grub/grub.conf add console=ttyS0 to
Anyone can request access to the wiki. Once you have access, contributions
can be made at will.
There is a forum at forum.sipfoundry.com that is tracking all of the emails
in this list.
At www.sipfoundry.com you can find the process for contributing code to the
community.
From: R
Hi Todd,
I've seen a lot documentations everywhere (sipxecs.blogspot.com,
myitdepartment.com) are only among the few. Maybe (just maybe) people can't
find a place where to put their contributions?
I know mailing list is here, but perhaps you'll all agree, it's hard to find
information in the maili
On Sun, 2010-04-18 at 18:12 +0800, Rhon wrote:
> Hi Everyone,
>
> I'm a little confused about the definition on Managed Phones and
> Gateways in SipXecs. Does the word "Managed" mean "Plug and Play"?
It means that there is at least some support for generating
configuration for them.
As others ha
Al sent an email out last week that listed every change to the software that
comprised the 4.2 release.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Francis Tinio
Sent: Sunday, April 18, 2010 9:43 AM
To: Picher
The wiki is completely built by community involvement. We all have an
obligation to fix what is not accurate, or create articles for the mutual
benefit of the rest of the community.
To quote Scott, "sipXecs is free, but not like beer". A pretty nice
analogy. Our cost as members of this commu
What is your seesion? Gnome? KDE? Terminal?
If terminal SSH. If xwindows (KDE or Gnome), use VNC. If you want KVM
to another PC, use an IP KVM system.
On Sun, Apr 18, 2010 at 1:01 PM, Charles Chalekson wrote:
> But if you reboot you get disconnected.
>
> How do you actually mirror the display ov
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
Says the last supported release is 2.1.3.
Antiquities don't play well. Replace the phones.
On Sun, Apr 18, 2010 at 1:01 PM, Josh Patten wrote:
> Unfortunately you're getting the bootrom and the firmware versions mixed up.
> The
Unfortunately you're getting the bootrom and the firmware versions
mixed up. The bootrom version supported on the IP 500 is 3.2.3revB and
the firmware version is 2.1.3. Both of these are very old and missing
some critical bug fixes and improvements.
In contrast on an IP 550 the bootrom version
Sorry, I forgot to include that you would still have to log into your system to
see the log. Or perhaps send all of your syslog output to another server.
On Sun, 18 Apr 2010 09:51:28 -0700, Charles wrote:
> As a linux novice running sipxecs on a machine that does not have a monitor
> attached I
You could output the console to syslog. Take a look at your syslog config in
/etc/
On Sun, 18 Apr 2010 09:51:28 -0700, Charles wrote:
> As a linux novice running sipxecs on a machine that does not have a monitor
> attached I had a question regarding monitoring the server.
>
> I frequently login
yup ip500s lol i wish they were at least 501s lol...
but hey they actually work. up tp 3.2.3 rev B :)
On Apr 18, 2010, at 12:54 PM, Josh Patten wrote:
> Am I to understand you are running IP500's? Not 501's or 550's?
>
> If so, it's amazing they actually work. The last firmware revision fo
http://www.lantronix.com/it-management/kvm-over-ip/securelinx-spider.html
Charles wrote:
> As a linux novice running sipxecs on a machine that does not have a monitor
> attached I had a question regarding monitoring the server.
>
> I frequently login via ssh/terminal to update/debug/etc the serve
if you are not directly connected, an ssh session is the easiest answer.
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: 434.984.84
Am I to understand you are running IP500's? Not 501's or 550's?
If so, it's amazing they actually work. The last firmware revision for
them was 2.1.3 which was known to have problems with sipX. They are
considered "End of Life" by Polycom and should be replaced ASAP.
Francis Tinio wrote:
o
www.sipfoindry.org
Article on page with 4.2 announcement has the link to the changes.
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
As a linux novice running sipxecs on a machine that does not have a monitor
attached I had a question regarding monitoring the server.
I frequently login via ssh/terminal to update/debug/etc the server, however, I
wanted to know if it is possible to mirror what would be seen on a monitor via
te
huh?
what you mean?
Also, where can I see the complete list of changes or new features when
upgrading 4.0.4 to 4.2? Couldn't quite find it in the site lol...
On Apr 18, 2010, at 12:37 PM, Picher, Michael wrote:
> Make sure though that the SIP helpers in the sonics are off...
>
>> -Origi
Make sure though that the SIP helpers in the sonics are off...
> -Original Message-
> From: Francis Tinio [mailto:fti...@toqen.com]
> Sent: Sunday, April 18, 2010 10:21 AM
> To: Picher, Michael
> Cc: sipXecs users
> Subject: Re: [sipx-users] cannot transfer between 2 remote locations
>
>
ok i got to update to 4.2 and got the polycoms to work with 3.2.3 bootrom.
there is no way for me to test if 4.2 will fix my issues with different
Locations trying to forward calls to each other.
I'll update you guys tomorrow once our client's office is open.
On Apr 18, 2010, at 11:05 AM, T
Also, I forgot to mention that when Location A calls Location B via extension
(not DID), the phone actually rings. If not picked up, the call goes to VM
successfully. But if picked up there is no audio.
On Apr 18, 2010, at 10:31 AM, Francis Tinio wrote:
> Both locations use one server that
I don't know your configuration, but the (new) wiki says to set up
each system as an unmanaged gateway for site-to-site calling.
In order to transfer the calls you need a site to site dialing plan .
What you are not saying is:
Whether or not you can natively dial between locations.
Whether or no
sonicwall, but ports are opened and calls actually work. I mean Location A can
take and receive calls, and forward calls within the same Location. The same
for Location B. The issue happens when Location A transfers to Location B or
Location A calls Location B.
On Apr 18, 2010, at 8:33 AM,
On Sun, Apr 18, 2010 at 9:22 AM, Scott Lawrence wrote:
> On Sat, 2010-04-17 at 21:20 -0400, Tony Graziano wrote:
>> On Sat, Apr 17, 2010 at 9:29 AM, Scott Lawrence wrote:
>> > On Sat, 2010-04-17 at 08:58 -0400, Tony Graziano wrote:
>> >>
>> >> Setting it up to dial directly "proxy-to-proxy" on po
On Sat, 2010-04-17 at 21:20 -0400, Tony Graziano wrote:
> On Sat, Apr 17, 2010 at 9:29 AM, Scott Lawrence wrote:
> > On Sat, 2010-04-17 at 08:58 -0400, Tony Graziano wrote:
> >>
> >> Setting it up to dial directly "proxy-to-proxy" on port 5060 does not
> >> work (no audio). Dialing via xlite sip u
On Sun, Apr 18, 2010 at 12:04 AM, Hiral Patel wrote:
> Hi,
>
>
>
> Can someone tell me what improvements have been made to sipXbridge in
> sipXecs 4.2 to stop all media anchoring on the sipXecs server?
>
>
>
> In sipXecs 4.0.4 we had an issue where if we used sipXbridge all media by
> default woul
>>> On 4/18/2010 at 5:53 AM, in message
, Rhon
wrote:
> Hi,
>
> I'm curious on how I can integrate Dim Dim with our SipXecs server. I have
> never seen any howto on how to accomplish this.
> Sorry for my ignorance, but can anyone give any idea on how to start?
>
> Here are some of my questions:
Tried it also... wiped the config files from the machine as well before
re-sending...
I've created a JIRA issue http://track.sipfoundry.org/browse/XX-8220
Thanks,
Mike
> -Original Message-
> From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> Sent: Sunday, April 18, 201
It's using a valid firmware as far as the phone is concerned. Looking
over the posts from other groups and searching with "5|00|
Compatiblity for image 0 is 0x11410." shows a lot of folks changing
the firmware version for that device. Would you consider uploading
3.1.6 and activating it and seeing
Sounds like maybe you have Internet Calling enabled... it should be
disabled.
Mike
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> boun...@list.sipfoundry.org] On Behalf Of Francis Tinio
> Sent: Friday, April 16, 2010 11:45 AM
> To: Scott Lawrenc
Ah, thanks.
> -Original Message-
> From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> Sent: Sunday, April 18, 2010 7:30 AM
> To: Picher, Michael
> Cc: Francis Tinio; sipx-users
> Subject: Re: [sipx-users] cannot transfer between 2 remote locations
>
> It was the ITSP. His trace s
It was the ITSP. His trace showed the ITSP sending a BYE in less that
20 seconds.
On Sun, Apr 18, 2010 at 7:29 AM, Picher, Michael
wrote:
> Francis, you should document on the list what you did to resolve your
> problems with the disconnect so that anybody else having the same type
> of problem c
Francis, you should document on the list what you did to resolve your
problems with the disconnect so that anybody else having the same type
of problem can have a good starting point... even if the issue isn't
sipXecs related (which I suspect it wasn't).
Mike
> -Original Message-
> From:
In your conference in sipxconfig, click web conference.
Dimdim Server(Default: webmeeting.dimdim.com)
Dimdim ID
If empty, the conference owner username will be used instead.
Dimdim password
Dial-in Number
Optional dial-in number that will be displayed
Hi Michael,
We always make sure that whenever we resolved issues, we post it on the
mailing list for those who might encounter the same problem in the future.
We will be documenting our project and will contribute to the community
whatever we've accomplished.
Best regards,
Rhon
On Sun, Apr 18,
I'm not sure if this is the case with all x01 Polycom models but the
601's are having some sort of serious error with the new 4.2 config
files. If you have any x01 series phones I'd hold off going to 4.2.
Here's the log from a boot. I'll open an issue in the tracker.
0418105018|so |4|00
The Polycom phones are the best supported phones. Some folks have been
working on some of the Cisco phones but they typically don't adhere
closely to the standards. Polycom remains the recommended phone and
x-Lite/eyebeam/Bria the recommended softphone.
Also on the gateway front, I don't thin
Hi Everyone,
I'm a little confused about the definition on Managed Phones and Gateways in
SipXecs. Does the word "Managed" mean "Plug and Play"?
Based in our experience with Cisco 7970G Phone and Audiocode MP118 (which
are both managed devices), these term does not fit well.
IMHO, the only devic
Hi,
I'm curious on how I can integrate Dim Dim with our SipXecs server. I have
never seen any howto on how to accomplish this.
Sorry for my ignorance, but can anyone give any idea on how to start?
Here are some of my questions:
1. The Community edition of DimDim is 4.5 but I've seen an API for D
40 matches
Mail list logo
