Good Day,
As far as I know, sipx looks through the dial plan list to find the first
match and if the user have the necessary permissions, it processes the call. If
the user does not have the proper permissions, the call is denied; and sipx
does not keep looking for a next best match.
Before I
Thanks again,
From: Picher, Michael mpic...@cmctechgroup.com
To: arda savran ardasav...@yahoo.com; sipx-users@list.sipfoundry.org
Sent: Sun, April 25, 2010 9:48:08 PM
Subject: RE: [sipx-users] how to construct a dial plan in an hosted environment.
Arda
We are trying to limit the resources that SIPX uses. Is there a way to limit
the total number of registered users on SIPX? or the total number of
simultaneous calls?
As far as I know, these things are doable on SCS500 through licensing. How
about SIPX? any workarounds?
Thanks
We currently have multiple ITSPs coming into SIPXBridge and we are using
internal sipXBridge to handle them. We can direct all inbound requests to
auto-attendant but what if we want to go a little bit more advanced than that.
Lets say we purchase 5 numbers from an ITSP and we would like 3 of
I am trying to subscribe a polycom soundpoint phone (phone B: extension 2002)
for MWI service for another extension (phone A) on sipx. My intensions are to
notify phone B as well when there is a new message waiting for phone A.
I built the configuration of polycom phone B on sipx. I put the
to their messages.
However like I said, user portal and MWI features go down..
Are you aware of this issue?
Thanks,
From: WORLEY, DALE R (DALE) dwor...@avaya.com
To: arda savran ardasav...@yahoo.com; sipx-users@list.sipfoundry.org
sipx-users@list.sipfoundry.org
Sent: Thu
for server A. If so,
what do you think the requirements are as far as configuration concerned on
both servers. (domain name, extension, dial plan etc.)
Thanks,
From: Picher, Michael mpic...@cmctechgroup.com
To: arda savran ardasav...@yahoo.com; WORLEY, DALE R (DALE
Can we still download 4.2 from sipfoundry? I cannot see it under available
downloads anymore. Does anyone know what happened?
Thanks
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive:
:
From: Picher, Michael mpic...@cmctechgroup.com
Subject: RE: [sipx-users] autoattendant issue with HA configuration
To: arda savran ardasav...@yahoo.com, sipx-users@list.sipfoundry.org
Date: Tuesday, December 22, 2009, 2:57 AM
Make sure there are no special characters in description
Good Day,
We have a cluster of 4 SIPX servers,
They are all connected to the same Layer2 switch and the system is operational.
The first two servers are running all the services except conferencing and
voicemail. The third server is conferencing and the fourth server is configured
as
, Michael mpic...@cmctechgroup.com wrote:
From: Picher, Michael mpic...@cmctechgroup.com
Subject: RE: [sipx-users] autoattendant issue with HA configuration
To: arda savran ardasav...@yahoo.com, sipx-users@list.sipfoundry.org
Date: Monday, December 21, 2009, 7:22 PM
I believe that VM must reside
: Picher, Michael mpic...@cmctechgroup.com
Subject: RE: [sipx-users] SIPX --- UDP port range for media services
andsignaling
To: arda savran ardasav...@yahoo.com, Tony Graziano
tgrazi...@myitdepartment.net, sipx-users@list.sipfoundry.org
Date: Tuesday, December 1, 2009, 11:58 AM
This doesn’t
Good Day,
I have been trying to cluster a couple of SIPX servers (4.0.2); but having the
same issue over and over again. I am wondering if anyone can point me to the
right direction,
I am using an external windows 2003 server as my DNS. My master SIPX server is
configured to use its local
I was wondering if there is any support on SIPX for the following features:
- Codec support (other than G.711u). Can SIPX support G.729 or G.711a?
- Can SIPX support Call Admission Control (CAC)?
- Can SIPX support QoS marking for SIP signaling and RTP packets?
Thanks in advance for all help,
Good Day,
I have an Audiocodes mediant 1000 box with some FXS ports. The rgistration on
the box is pointing to SIPX and my analog phones that are connected to
Audiocodes, are registering to SIPX.
I can see the registrations on SIPX for the analog phones as the following:
URI:
I am desperately looking for an answer for this...Any help is most
appreciated...
We are trying to put together a remote office office solution with SIPX located
at a main location and SIP clients located at the remote locations.
Each remote location is going to have an Audiocodes Mediant 1K
Good Day,
I was wondering if anybody has ever tried connecting an audiocodes mediant 1000
box to a SIPx to try Audiocode's remote surviability feature (SAS).
In my test network everything including SIPX, DNS server, Audiocodes Mediant 1K
and my sip phones (cisco 7940 and polycom sip
I see many different templates available for gateways in SIPXecs. I was
wondering if it is possible to add a custom template for a gateway (for
example, an ADTRAN voip gateway) into this list.
I do not mean the custom gateway option that SIPXecs provides in the drop
down menu. What I mean is
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
pattern should match the numbers used on
CCM, the gateway should point to your trunk and only “Local Dialing” permission
should be required.
Regards,
Gabor
From: arda savran [mailto:ardasav...@hotmail.com]
Sent: 01 July 2009 18:03
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users
I built a SIP trunk between a CUCM6 and SIPXECS server in our lab.
Everything in CUCM6 regarding SIP trunk is set to non-secure standard.
I can make phone calls from the phones registered to the sipxecs server to the
phones registered to CUCM6 server.
However, when I try the the other way
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