Here and therefore would
not ring when there is an active call for user 200 and a call is placed to
the hunt group.
Sincerely,
Dave Deutschman
Sr. Director, Customer Services
eZuce, Inc.
Office: (978) 296-1005 x 2037
Mobile: (425) 478-9642
-Original Message-
From: sipx-users-boun
:
https://www.ezuce.com/jobs
If anyone is interested, please contact me off list.
Let's keep this community growing!
Sincerely,
Dave Deutschman
Sr. Director, Customer Services
eZuce, Inc.
Office: (978) 296-1005x 2037
Mobile: (425) 478
Douglas,
I can confirm that in 4.4, if a phone had the Do Not Disturb button pressed
and the person had a voicemail account, that the system would fork the call
to voicemail when the 486 Busy Here message was received from the phone.
DD
From: sipx-users-boun...@list.sipfoundry.org
[mail
I have also seen this on a 4.4 system.
Calls were not being processed and the caller heard dead air. When a call
was researched, I found that sipXproxy was loading the NAT plugin for the
call even though the call was outbound to the PSTN via an AudioCodes
gateway.On this system, NAT trav
case) then authentication works when a user types in their
alias because sipx maps the alias to the username and sends the
username to LDAP and LDAP authenticates it against IPPhone.
Kyle
On Thu, Aug 16, 2012 at 8:50 AM, Dave Deutschman
wrote:
> I would like someone to verify LDAP functionality
e searches also based on aliases as
well, so basically if a user has 3 aliases we would need to make 1 to 4
searches in ldap until the user gets authenticated...
Mircea
Dave Deutschman
Managing Partner
Innovational IP Solutions, LLC
PO Box 983
Bothell, WA 98041
206.965.9586
validate using the LDAP credentials?
Dave Deutschman
Managing Partner
Innovational IP Solutions, LLC
PO Box 983
Bothell, WA 98041
206.965.9586 x 301 (o)
425.478.9642 (m)
<mailto:3...@sipx.isdomain.net> 3...@sipx.isdomain.net (s)
<http://www.innovational.net/> www.inno
.
There was a bug in the startup scripts of 4.2.0 where it started the process
on a secondary but that had been fixed in a 4.2.1 patch release by Nortel.
I am not sure if that fix made it back to the open source builds.
Sincerely,
Dave Deutschman
Managing Partner
Innovational IP Solutions, LLC
Todd,
I am not aware of a plug-in. We have developed SQL scripts that extract the
CDRs from postgres and upload into another database or call accounting
package. Many of the call accounting packages have tools to import data.
The issue is that you have to extract the extensions/numbers from th
Roman,
sipXecs does not perform any dips into a CNAM database. On an inbound call,
it passes whatever display name content is included in the URI by the SIP
Trunking provider or is constructed by the media gateway from the LEC/CLEC
ISDN messages to the phone.
On an outbound or internal call, it
Sven,
The value is an Advanced Setting of the SIP Registrar service.
DD
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Sven Evensen
Sent: Tuesday, January 10, 2012 7:08 AM
To: Discussion list for users of sipXecs software
Subje
I think everyone has made their point and it is time to move on.
Thanks to all who contributed to the project in 2011. Hopefully we can
expand that list in 2012 and make this an even better project!
Sincerely,
Dave Deutschman
-Original Message-
From: sipx-users-boun
Calling Name for traditional land line and wireless carriers is determined
by a database lookup by the carrier terminating the call. It is based on
the Caller ID. It is a fee based service in which both the originating and
terminating carriers must agree to participate. Almost all land line
carr
We have found that you can use BootROM 4.2.1 instead of 4.3.0 and that fixes
the problem.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Adrien Guillon
Sent: Wednesday, October 19, 2011 3:20 PM
To: sipx-users@lis
It also is not responding when I attempted to connect to the home page.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel
Watkins
Sent: Wednesday, August 17, 2011 1:58 PM
To: Discussion list for users of sipXecs software
Subject:
User information is stored in postgresql in the database SIPXCONFIG. The
data is contained in multiple tables.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Paul Curtis
Sent: Tuesday, July 12, 2011 3:19 PM
To: sipx-users@list.sipfound
users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Dave Deutschman
Sent: 14 June 2011 16:14
To: 'Discussion list for users of sipXecs software'
Subject: Re: [sipx-users] Using SIP Viewer on older files
Sven,
Use the -e option (extension). The extension
Sven,
Use the -e option (extension). The extension needs to be enclosed in a
single quote.
Example
sipx-trace -a -e '.1' -o test.xml
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Sven Evensen
Sent: Tuesday, June 14, 201
+1
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitchin
(public/usenet)
Sent: Wednesday, May 11, 2011 6:26 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Bug fix release
+1
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, May 05, 2011 6:31 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] XX-9565 voicemail recording - feedback needed
yeah if i
What bootrom are you using? We have seen this condition occur with
SoundStation IP 6000s when using bootrom 4.3 with sip.ld 3.2.4 or 3.2.5. It
does not occur if you drop back to bootrom 4.2.2.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Beha
There are two Advanced settings on the Grandstream that can affect attended
transfers. The first is Enable Call Features. It must be set to Yes. The
second is Disable Call-Waiting. It has been some time since I have worked
with the phone, but I believe that option needs to be set to No.
I
There is a bug fix in 4.2.1 that broke directed call pickup.
To enable pick-up to aliases and huntgroups:
1. Add
SIP_REDIRECT.100-PICKUP.ENABLE_XX_8438_WORKAROUND : Y
to
/etc/sipxpbx/sipxregistrar/registrar-config.vm
2. Go to System/Servers//SIP Registrar and cl
sipXecs issues a NOTIFY message with the Event: element containing the value
check-sync. That is the method implemented by most IP phones.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jeff Gilmore
Sent: Friday, March 11, 2011 5:43 AM
There are two issues with their SIP DECT wireless system running Aatra Open
Mobility Manager V 1.8.9 and 620d handsets.
1. When you attempt to terminate a call from the PSTN, it will hang an
analog port on an AudioCodes gateway. The issue appears to be that the
Aastra OMM software is sw
We have been using Ingate SIParators for some time. They have been
certified with most SIP Trunking providers and sipXecs. Their latest
release includes an IDT/IPT module that kills sipvicious attacks at the
firewall.
The only short coming we have found is that their software does not fully
s
Todd,
The data is stored in the database. There are two tables that would contain
the phone information:
phone
discovered_devices
The tables are linked by phone.serial_number and
discovered_devices.mac_address.
You can use a tool or psql to delete the rows. You would need to fi
Todd,
If the Request URI contains an IP address instead of the sip domain for
sipXecs, do you have the IP address configured as a Domain Alias? If not,
that would be why sipXecs is issuing the 404 Not found.
DD
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@lis
Polycom just provided sip.ld 3.2.4 to some of its' partners including Avaya.
We have it installed at one customer with SCS 4.0 systems who was using
3.2.3 sip.ld in support of Shared Line Appearance and encountering the "one
ring, role to voicemail" issue caused by the phone issuing a 400 Bad Reque
If all services are running and all function working, that tells me all is
OK.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
m...@grounded.net
Sent: Wednesday, January 12, 2011 2:29 PM
To: sipx-users
Subject: Re: [sipx-users] Changing
Behalf Of
m...@grounded.net
Sent: Wednesday, January 12, 2011 2:13 PM
To: sipx-users
Subject: Re: [sipx-users] Changing IP of a live server
On Wed, 12 Jan 2011 13:57:38 -0800, Dave Deutschman wrote:
> You never performed Send Profile for the server after changing the IP
> address in step 10
You never performed "Send Profile" for the server after changing the IP
address in step 10. You need to perform this step to regenerate the config
files as the data is stored in postgres. That is why the config files are
wrong.
Also, running grep against the postgres library is not a valid st
If all the steps are followed in the correct order, you can change the IP
address to any valid value, including a former IP address.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
m...@grounded.net
Sent: Tuesday, January 11, 2011 7:26 P
After changing the IP address on the config page, you always need to Send
Profiles as that generates the config files used by all of the processes.
You may have entered the IP address as an alias of the domain. If so, you
need to change it also on the System / Domain page. After making a change
To check the value in the database, at the command line enter:
psql -U postgres -d SIPXCONFIG
Then at the prompt enter:
select location_id, name, ip_address from location;
Verify the IP address in the column ip_address.
Enter \q to exit psql
This IP address is managed on th
Double-check the column ip_address for the table location in the postgres
database SIPXCONFIG. Make sure it reflects the value of your new IP
address. If it does not and you change the value, you will need to Send
Profiles for the server(s) to generate new configuration files.
From: sipx-user
The way that sipXecs works is that when sipXproxy receives the 302 Moved
Temporarily with the list of phones that are to receive INVITES from
sipregistrar, it starts issuing the INVITEs based on the q value. If you
have a large number of phones with the same q value, be it the same
extension or di
These questions may have been asked and answered before, but I could not
find them in the archives.
What is the tested maximum number of call control servers in a distributed
system? Based on the following paragraph from the Wiki site, it sounds like
we can have 4 or more, but has that configura
requirements of the customer.
Sincerely,
Dave Deutschman
Managing Partner
Innovational IP Solutions, LLC
PO Box 983
Bothell, WA 98011
206.965.9586 x 301 (o)
425.478.9642 (m)
<mailto:3...@sipx.isdomain.net> 3...@sipx.isdomain.net (s)
www.innovational.net <http://www.innovatio
If it is disabled by default, then value that is supplied in the attribute
"msg.mwi.1.callBack=" must have the domain stripped off as in v3.10.3, it
contains the domain name. Disabling the feature and using the domain name
in the value for msg.mwi.1.callBack= breaks the ability to dial voicemail
rated configuration file is indeed the correct
version for sip.ld v2.2.2?
Sincerely,
Dave Deutschman
Managing Partner
Innovational IP Solutions, LLC
PO Box 983
Bothell, WA 98011
206.965.9586 x 301 (o)
425.478.9642 (m)
<mailto:[EMAIL PROTECTED]> [EMAIL PROTECTED] (s)
www.innovationa
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