That could be quite a read!
Microsoft is developing... = We haven't even started this product yet
but we want to stifle the competition and then just buy some company.
Microsoft XYZ Version 1 = This is really going to suck, but we want
to get something out the door.
Microsoft XYZ Version 2 =
Does bind run on that box? If so, and it answers for its own zone, look at the
ZONE file itself. If not, verify the entries for the zone file are in the bind
zone file that is authoritive to the sipx system having the issue for that
zone. \
If you are being hit by this:
Is there a way to call from PSTN, enter a PIN (or without one) and dial any
number, supported by dialplan?
Security may be used at any level:
- PIN
- CID of incoming caller
- PSTN gateway
I found only one way (using autoattendant), but this method is limited by
preprogrammed numbers.
No.
-Original Message-
From: s...@pharmacia.org.ua
To: sipx-users@list.sipfoundry.org
Sent: 7/18/2009 10:25:14 AM
Subject: [sipx-users] international calls from PSTN through SipX
Is there a way to call from PSTN, enter a PIN (or without one) and dial any
number, supported by dialplan?
hi Kevin
Here is the sipx-dns advisor dump
r
a.. From: Kevin Thorley kevin.thor...@xx
b.. Date: Fri, 17 Jul 2009 18:48:30 + (UTC)
On Fri, 17 Jul 2009 13:35:22 -0500, voice wrote:
Thanks Scott
Looks like i can't send attachment
he is the output from the sipx-dns advisor
;
; ! Missing DNS records:
;
; NAPTR record for chicagonettelephone.org
;
Your zone file looks empty. See the last message I sent with the associated
tracker item. You'll need to re-enter some records manually to fix it. I'd
suggest once it is fixed telling bind to freeze the zone and maybe it won't be
overwritten again.
-Original Message-
From: robertNGN
ON a more serious note, Microsoft has made a substantial investment in a
core switch platform, from Nortel. There is a good sized Nortel team
assigned specifically to Microsoft to help with development of the OCS
platform, and Microsoft is making a serious investment of talent to develop
their
Hi,
I have a problem that I need help with. When I call in the Auto Attendant
responds without any problem, but the call is dropped soon after I dial the
extension (200). With that said any calls from the inside go out without a
problem, which would mean that the Gateway is working just
It sounds like the gateway is not handling the refer properly. I don't use
audiocodes anymore, but that kind of behavior can be other things. Make sure
you have the correct or recent firmware. You should be able to get some help
from this list because a lot of people here use audiocodes too.
Dear All,
I am planning to upgrade a few installations running on 3.10.2 to latest
4.x (what is the latest-known-to-be-running-well version btw?) via yum.
I've tried installing a test system with the latest 4.0.1 centos iso few
weeks ago but that throw error 'sipxecs package cant be located'
Thank you Tony
i used my sipx clone's zone file for sipx1 and everything back to normal
problems and all. This issue came about while trying to solve why i can't
create a server for sipxBridge.from the system/servers/add server.
*** I may have found the root of the problem ***. when i try to
The easiest way to debug this:
- Enable debug level 5 on AC Gateway.
- Go to Message log screen on the AC GW and then make an inbound call
through Attendant. The message log should show the request sent and response
received which should give enough info to debug this.
svs
- Original Message
Overlapping ZONE issues
Thank you Tony
i used my sipx clone's zone file for sipx1 and everything back to normal
problems and all. This issue came about while trying to solve why i can't
create a server for sipxBridge.from the system/servers/add server.
*** I may have found the root of the
Hi Cuneyt,
I would suggest making a copy of the ZONE file. I actually just paste mine into
a google doc for safekeeping.
Other than that you can follow the wiki and it will work fine. If you ZONE file
is emptied after the reboot, just copy the file back over and restart named and
sipxecs.
I'm having trouble with a 3.10.3 installation.
When I only enable Exchange integrated voicemail, it works properly.
However, if I create a custom Dail plan for Exchange Autoattendant, it breaks
voicemail.
The ExchangeAA dialplan simply has an extension of 999 and send it to a
configured
I have been trying to cluster a couple of SIPx server. When I check the master
unit, I can see the secondary unit registered to it with the following services
up and running:
1)CDR HA tunnel
2)Media Services
3)SIP Trunking
4)Media Relay
5)SIP Registrar
6)SIP Proxy
However I can not make any
Feature is called DISA. Not allowed.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
s...@pharmacia.org.ua
Sent: Saturday, July 18, 2009 10:25 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users]
They can do whatever they set their minds to... no doubt about that.
Fits and starts however don't breed a lot of confidence in customers.
I think OCS is 'getting there' but it is basically version 2 of Live
Communications Server. It might be close to right next go-around. By
the third release
I have serious problems with the dual line telephones, if I have an
active call and a new call comes, I cut the incoming call. I have
installed 3.10.2 with Polycom 320/330 Phones with a AUDIOCODES M1000 I
have installed this version and the version 4.0.1 causing similar
problems and I thought
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