Hi All,
please send profiles to server after updating to latest 4.6 RPMs -
there was a change in conferences records so they need to be
regenerated
Thanks
George
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On Thu, Oct 11, 2012 at 3:58 AM, George Niculae geo...@ezuce.com wrote:
please send profiles to server after updating to latest 4.6 RPMs -
there was a change in conferences records so they need to be
regenerated
George,
When sipxconfig restarts, it regenerates profiles but does not rebuild
On Thu, Oct 11, 2012 at 12:05 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Thu, Oct 11, 2012 at 3:58 AM, George Niculae geo...@ezuce.com wrote:
please send profiles to server after updating to latest 4.6 RPMs -
there was a change in conferences records so they need to be
regenerated
On Thu, Oct 11, 2012 at 5:16 AM, George Niculae geo...@ezuce.com wrote:
On Thu, Oct 11, 2012 at 12:05 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Thu, Oct 11, 2012 at 3:58 AM, George Niculae geo...@ezuce.com wrote:
please send profiles to server after updating to latest 4.6 RPMs -
there was
On Thu, Oct 11, 2012 at 5:16 AM, George Niculae geo...@ezuce.com wrote:
On Thu, Oct 11, 2012 at 12:05 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Thu, Oct 11, 2012 at 3:58 AM, George Niculae geo...@ezuce.com wrote:
please send profiles to server after updating to latest 4.6 RPMs -
there was
I am a total newbie on SipXecs. I am also green when it comes to the SIP and
VoIP PBX scene. Please excuse my seemingly simple question.
The problem that I am encountering, essentially, is that external calls cannot
be transferred to voice mail when a call is not answered. Internal calls
On 10/11/2012 10:37 AM, Henry Kwan wrote:
I am a total newbie on SipXecs. I am also green when it comes to the
SIP and VoIP PBX scene. Please excuse my seemingly simple question.
The problem that I am encountering, essentially, is that external
calls cannot be transferred to voice mail when a
I don't think the router is compatible with the ability to 1:1 NAT or
do NAT without changing (randomizing) the source port. I would get
thee to a router that will do thusly. Even if you do all of the above,
you will likely have frequent or all the time broken audio.
On Thu, Oct 11, 2012 at 10:37
Hi,
I'm not sure if this is a known bug. I've installed SipX twice now, and
upon booting the screen is completely garbled. I suspect a framebuffer or
something is being used for boot, but it doesn't like my hardware. Anyone
else experience this?
AJ
Realize you have not stated what version of sip and how you are
installing it. Is it sipx 4.4 or 4.6? Are you installing via RPM or
from ISO?
Typically it means your hardware has an issue with linux. If you know
what hardware you are using (we don't) and what version of sipx you
are installing
It's an ISO install, 4.6.
Yes, it could be an issue with my hardware and CentOS, however the issue is
that not all hardware is going to support framebuffers properly.
Regardless of whether the bug is in CentOS or the install scripts of the
sipxecs ISO, it is still an issue that there is no option
Well, you could use the RPM installation method and go that route... Then
you'd have full control over the installation.
Thanks,
Mike
On Thu, Oct 11, 2012 at 11:56 AM, Adrien Guillon aj.guil...@gmail.comwrote:
It's an ISO install, 4.6.
Yes, it could be an issue with my hardware and
Do I need one-to-one NAT, or symmetric NAT? I bought this router because it
had been tested to work with VoIP, whatever that means, but I forgot the source
of this information.
From: Tony Graziano tgrazi...@myitdepartment.net
To: Henry Kwan hslk...@yahoo.ca;
I find that 1:1 is best with access rules only allowing the ports you want,
this way server always goes out with the same IP. Also, make sure the
firewall does not do outbound port randomization.
On Thu, Oct 11, 2012 at 12:18 PM, Henry Kwan hslk...@yahoo.ca wrote:
Do I need one-to-one NAT, or
I'll add one more thing that works...
On the Windows version of Bria 3.x (not the mac, linux, or any of the
tablet/iphone/android) there is the ability to setup RLS. You can enter
your RLS subscription and allow other phones to monitor your softphone's
presence. For a workgroup server enter:
On Thu, Oct 11, 2012 at 11:56 AM, Adrien Guillon aj.guil...@gmail.com wrote:
It's an ISO install, 4.6.
Yes, it could be an issue with my hardware and CentOS, however the issue is
that not all hardware is going to support framebuffers properly. Regardless
of whether the bug is in CentOS or
On Thu, Oct 11, 2012 at 12:18 PM, Henry Kwan hslk...@yahoo.ca wrote:
Do I need one-to-one NAT, or symmetric NAT? I bought this router because it
had been tested to work with VoIP, whatever that means, but I forgot the
source of this information.
Disable SIP algorithm if it's enabled router.
It's a centos issue.
It's probably a resolution issue. it's still centos and your hardware
let me google that FOR you...
http://lmgtfy.com/?q=framebuffers+centos+6
the answer lies in probably changing grub.conf to match what the
hardware might support...
example:
kernel
Tested by who? Just because it works as a home router for voip doesn't
mean it will probably work for your office hosting a PBX, BIG FAT
difference.
On Thu, Oct 11, 2012 at 12:18 PM, Henry Kwan hslk...@yahoo.ca wrote:
Do I need one-to-one NAT, or symmetric NAT? I bought this router because it
I'll download a minimal CentOS install now to try to help narrow down the
problem to CentOS or SipX. Thanks for pointing out the potential install
parameters, if X runs during the install (which it did) I would assume that
the framebuffer would be alright.
Tony: I am well aware of how to change
Okay, so I tried to CentOS 6.3 minimal install. The issue remains with a
normal install, however CentOS also has a low-end video mode, and that
install works just fine.
It might be worth considering, if simple enough, to add an option to SipX
to install CentOS in a low video mode. For now I can
The router, Linksys WRVS4400N, that I am using is not a home router. It is a
small business router. Having said that it still may not mean it is a suitable
router for SipX.
I managed to obtain another router and do more testing tonight. The router is
a Linksys/Cisco RV016. It has
All,
I just realized that my emails from my SipXecs 4.4 server were not being
delivered. Upon further investigation, I found that my SipXecs VM had a
sendmail queue with over 13000 messages in it. I'm trying to figure out how my
machine was sending mail, and it doesn't look like the relay is
Hi, could be something related to Polycom's phones FTP provisioning ? I've
read that the default FTP user name for that is 'PlcmSpIp' and the default
password is the same (so well-known credentials).
Over ther internet there are some references about that (AFAIK see
this
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