I found the answer to this problem by accident today. The new Avaya
phones get their caller id information from the 'Contact' header.
Interestingly, the ACD function of sipX uses full contact header
information, and thus, caller id displays on Avaya phones.
Any chance there is an easy way to cha
That's correct as I understand it, HA doesn't work unless PROXY role
is selected on ANY of the members.
Feel free to solicit other opinions.
On Fri, Sep 3, 2010 at 2:57 PM, McIlvin, Don
wrote:
> To help clarify (Pete and I work together)
>
>
>
> The HA Server
>
> CDR HA Tunnel Running
>
> Shar
To help clarify (Pete and I work together)
The HA Server
CDR HA Tunnel Running
Shared Appearance Agent Running
Media Relay Running
SIP Registrar Running
SIP Proxy Running
The Bridge Server
SIP Trunking Running
The Primary server
has everything, except SIP Tru
Sorry. I think in 4.0 stdprompts was where they were... now that the
media server is on FS thats handled differently. I cannot find any
"standard.wav" file in 4.2x so it must be wrapped up in a java archive
and pulled out when a new user is created.
I will see if I can find out where...
On Fri, S
On Fri, Sep 3, 2010 at 2:20 PM, Worley, Dale R (Dale) wrote:
>
> From: sipx-users-boun...@list.sipfoundry.org
> [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Talbot, Peter
> [peter.tal...@nrtnortheast.com]
>
> shouldn’t SBC1 be trying to send that
On Fri, Sep 3, 2010 at 2:20 PM, Worley, Dale R (Dale) wrote:
>
> From: sipx-users-boun...@list.sipfoundry.org
> [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Talbot, Peter
> [peter.tal...@nrtnortheast.com]
>
> shouldn’t SBC1 be trying to send that
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Talbot, Peter
[peter.tal...@nrtnortheast.com]
shouldn’t SBC1 be trying to send that traffic off to SIPX1 based on DNS SRV
Records?
Hello,
Can someone point me in the right direction to where I can location the
default sipXecs 4.2 greeting wav files, I have looked under stdprompts
and it is not there.
For some reason when we change the user to the "standard" greeting form
the system default greeting there is no change,
Attempting to get HA working with sipX 4.2.1 and running into problems.
We are currently running on 3 SIP servers:
SIPX1 - Primary SIP Server with all Roles except SIP Trunking
SIPX2 - Redundant SIP Router role only
SBC1 - SIP Trunking (Primary SIP Gateway to ITSP) role only
Our DNS SIP
ctrl-Insert might be more linux fiendly???
Paul
>
> Of course it might be possible to send "ctrl&c" to the active window
> when pressing the global hotkey.
> But this is a bad idea if the active window is a linux shell session...
>
> There are some ways to skip this step. E.g. we implemented a
Of course it might be possible to send "ctrl&c" to the active window when
pressing the global hotkey.
But this is a bad idea if the active window is a linux shell session...
There are some ways to skip this step. E.g. we implemented a click to dial
solution for a callcenter where agents simply sel
René
Clever implementation - is there a way to do away with the clipboard in the
middle - if your app is there listening - could you simply have the number
selected and press the hotkey? Worst case - you could send keys to the active
window and put it in the clipboard - seems like the clipboar
There are quite a few attachments that got lost in the transition from Avaya
as we switched servers. They show the error you found below. This has no
affect on the system, it just means the file is not there.
--martin
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun
There are quite a few attachments that got lost in the transition from Avaya
as we switched servers. They show the error you found below. This has no
affect on the system, it just means the file is not there.
--martin
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun
We have created a simple tool for doing click to dial out of any application
using the clipboard with sipx.
The steps are:
- Copy number to clipboard
- press a hotkey
- a small window opens where you can modify the number that shall be dialed
- press okay and click to dial is started
We planned to
the other one that made sense to me was called "snap-a-number". The
developer had some issues, he also sold it to digium. Digium rebranded
it "ADA" (Asterisk Desktop Assistant) which is pretty to look at but
it is not "sip" it is "asterisk. Besides the huge memory issues with
the application, neith
I was looking at a few issues, and on all of them I get a "system error":
Oops - an error has occurred
System Error
A system error has occurred.
If this problem persists - please notify your JIRA administrator of this
problem.
See for example the attachments of:
http://track.sipfoundry.org/browse/
A couple of thoughts...
1. It would have to be written as an installable package for the OS.
2. Why would it create a link to launch click to call when it could
simply send the invite to the system for you which would then ring
your phone to complete the call?
3. It would have to consume a fair am
We have the hardphone acting as a switch between the network outlook and the
computer. Click to call works great via the web portal. I would like to know
if its possible to (without a softphone):
1. Convert all telephone numbers found on a web page to click to call
2. Create a href link that
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