From: Nathan Nieblas [nathan.nieb...@sacatech.com]
Please forgive my ignorance but I only see yum repo's for stable and
devel but nothing in between. It doesn't look like the version it's
fixed in has been committed to stable - how would I update using yum
I’m sure someone will correct me here if I am wrong but, it is my
understanding that the scenario described with the ITSP below isn’t
possible at this time. I believe that three issues are at work: NAT
translation, authentication, and ITSP support for re-invite (this was
discussed in the past, se
Hello guys,
I'm facing a weird issue.
On a new installation a SCS 3.0 (sipx 4.04) is answering to the audiocode INVITE
with a "407 proxy authentication required". SCS and audiocode are on LAN.
The matching rule on dialplan should call a autoattendant.
I Never seen this beforewhy should S
This is on an IIS/PHP/Curl box. User sweeks is not on our voip system - so
I've added a 'phantom user' and enabled call forwarding to reach our old phone
system via an analog gateway. Running this php code will contact his NEC phone
and ring my extension.
Our plan is to query a database with
Wow - that was ridiculously easy...
Thanks again :)
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel Watkins
Sent: Friday, May 28, 2010 9:49 AM
To: M. Ranganathan
Cc: sipXecs users
Subject: Re: [sipx-users] Click to call from our
Perfect - It's running on a different web server - I'll check out the examples
- and try the correct port.
Thanks
Nathaniel Watkins
IT Director
Garrett County Government
203 South 4th Street, Room 210
Oakland, MD 21550
Telephone: 301-334-5001
Fax: 301-334-5021
E-mail: nwatk...@garrettcounty.org
If you are calling from a trusted host ( i.e. one where sipx proxy is
running) please HTTPS (no authentication is required).
If you are calling from a different host, then you must authenticate with
Digest authentication.
There are some examples with notes in the source directory.
sipXcallControll
I would like to enable click to dial from our intranet site. I'm attempting to
follow:
http://wiki.sipfoundry.org/display/xecsdev/SipXCallController+REST+service
I have yet to be successful. I've attempted various methods (simply putting
the appropriate data into an html form and having
'htt
I had a 3.10 with external authrules and fallbackrules and upgraded to 4.0
without any issue.
Not sure about the external aliases, but I think it will go well.
When the time comes, I'll try an upgrade to 4.2.
-
MM
On Fri, May 28, 2010 at 10:02, Pete Burgess wrote:
> Thanks for the opinions. I
Thanks for the opinions. I'll follow the recommended upgrade path.
Thanks
Pete
-Original Message-
From: Douglas Hubler [mailto:dhub...@ezuce.com]
Sent: Friday, May 28, 2010 7:58 AM
To: Tony Graziano
Cc: Pete Burgess; sipXecs users
Subject: Re: [sipx-users] 3.10.2 to 4.2 with External A
On Fri, May 28, 2010 at 5:54 AM, Tony Graziano
wrote:
> I'd strongly urge 4.0 first. There are schema changes in 4.0 and 4.2. 4.2
> schema changes do not go back to 3.10x, so 2 steps is necessary.
> No also on the aliases.
I'm fairly sure the database knows how to go from any earlier version
to t
Hi
Is it possible in SIPXecs 4.2 to restrict the number of calls to a remote
site (e.g. branch office)? And also to restrict the number of calls per
individual user?
Similar to Locations in Cisco CallManager, where you can specify how much
bandwidth is available for voice at a remote site, and t
thanks,
I'll try it out an let you know.
It matters if you have (theoretical speaking) the remote user trying to call
an outside number, then the media goes like this :
remote user internet - sipx -- internet -- voip provider
and i want the media to go directly to provider (beca
The only thing you can try is aggressive mode in sip to try to see if the
media will establish as peer-to-peer.
Remote user utilizes media relay, has 2 modes (conservative & aggressive).
It requires nat be setup at your router properly for either to work.
I don't understand why it matters, if the
ok ok.. i think we don't understand each other completely..
I understand that 3cx and sipx are different. I have put 3cx just as an
example to describe to you what I want to do.. I don't even know if that can
be done with sipx. That's what I'm trying to find out. And that's why I have
asked you al
I'd strongly urge 4.0 first. There are schema changes in 4.0 and 4.2. 4.2
schema changes do not go back to 3.10x, so 2 steps is necessary.
No also on the aliases.
On Thu, May 27, 2010 at 12:46 PM, Pete Burgess wrote:
> Hi folks
>
>
>
> From looking at some of the docs it looks like I can upgra
I've always done a 4.0 as an intermediate step.
Have not done it with any external aliases.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Pete
Burgess
Sent: Thursday, May 27, 2010 12:47 PM
To: 'sipXecs users'
Subject: [sip
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