I reread your post, i totally misunderstood. I thought you were ready
to make a concession for now until bria support for redundant, tls
servers fix is committed.
ezuce wants tls to work on redundant systems so we'd have a vested
interest in getting this fixed in 4.6.0, it's only a matter of time
Hi!
I have a problem with the call transfers.
I'm using sipXecs (4.4.0- 2012-09-29).
I can call the AA. (working)
I can call an extension. (working)
I can call an external phone. (working)
When I call the AA and the AA transfer this call an extension I hear this:
Please hold while i transfer
On 10/5/2012 12:12 PM, Douglas Hubler wrote:
Update #20
==
- ** No security updates in this update **
- ISO has *not* been rebuilt as decided in release policy. Yum update
after installation is recommended for getting these updates.
On 10/10/2012 9:07 AM, Veréb Norbert wrote:
Hi!
I have a problem with the call transfers.
I'm using sipXecs (4.4.0- 2012-09-29).
I can call the AA. (working)
I can call an extension. (working)
I can call an external phone. (working)
When I call the AA and the AA transfer this call an
On 2012.10.10., at 15:16, Gerald Drouillard wrote:
On 10/10/2012 9:07 AM, Veréb Norbert wrote:
Hi!
I have a problem with the call transfers.
I'm using sipXecs (4.4.0- 2012-09-29).
I can call the AA. (working)
I can call an extension. (working)
I can call an external phone. (working)
I am trying to to track down an issue where a site-to-site dialing plan is
not working as expected.
The plan lets one site dial the other site auto attendant BUT not any of
the users directly. The call is stopped at the local proxy with a 404 not
found and it never gets passed the proxy
describe the call coming in. the invite (if from an itsp using siptrunking
built-in) must be sent to port 5080, otherwise the REFER will fail.
On Wed, Oct 10, 2012 at 9:31 AM, Veréb Norbert vereb.norb...@webcode.huwrote:
On 2012.10.10., at 15:16, Gerald Drouillard wrote:
On 10/10/2012 9:07
When I call the AA from the other extension, the problem is the same. But I'm
just calling the hung group the transfer is working.
On 2012.10.10., at 15:35, Tony Graziano wrote:
describe the call coming in. the invite (if from an itsp using siptrunking
built-in) must be sent to port 5080,
It might be that your user agent (phone) is either not configured properly
or does not support REFER. You still haven't provided any real information.
User agent
how is it registering (local/remote)
if remote, describe the firewall configuration
On Wed, Oct 10, 2012 at 9:44 AM, Veréb Norbert
G
Thank you! It's solved. The user agent phone doesn't support REFER.
On 2012.10.10., at 15:50, Tony Graziano wrote:
It might be that your user agent (phone) is either not configured properly or
does not support REFER. You still haven't provided any real information.
User agent
Ugh. nevermind. The gateway was using IP but the one system didn't have its
ip as an alias (which is the default in 4.6)!
On Wed, Oct 10, 2012 at 9:34 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
I am trying to to track down an issue where a site-to-site dialing plan is
not working
I have 2 snom 370 phones with expansion bases connected to sipXecs 4.6 working
properly. They will monitor and display the presence of users that have Polycom
phones however none of the softphone users can be monitored from the snom
phones.
I'ved tried several different softphones but no luck.
You need at least Bria 3.x. None of the other phones support the
methods needed. The phone presence and xmpp need to be integrated into
the same software package (bria 3.x), the polycom phone can only be
monitored from a softphone if the polycom user is also running a
proper version of xmpp.
See
Thanks Tony. I got too hung up on it being a snom/softphone compatibility issue
to step back and look and the rest of the situation.
Thank you for the second set of eyes!
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
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