ot accept
calls not from my itsp" would be one method to help control this.
-M
>>> "Todd Hodgen" 08/07/10 12:47 PM >>>
There is an analogy that works well here. Today, you can call any telephone
number you want, ring the phone and hang up. This isn'
Congratulations Martin, Jerry, Mike, Doug and the rest of the team. It's an
exciting day for all of you!
The site looks great, love the graphics! Best of luck to eZuce.
From: sipx-dev-boun...@list.sipfoundry.org
[mailto:sipx-dev-boun...@list.sipfoundry.org] On Behalf Of Martin Steinmann
S
There is an analogy that works well here. Today, you can call any telephone
number you want, ring the phone and hang up. This isn't much different, a
user can use sip to call directly into a sip phone. And, as kids I think
many of us can recall playing pranks on people over the phone - caller I
I think you have to look at the logic behind what is supported. Products
that weren't strategic to a Pingtel or Avaya sales and marketing effort
didn't have a lot of chance of being put into the product by them. So,
unless someone provides a solution that they have developed and put into the
proj
, 2010 2:22 PM
To: 'Todd Hodgen'
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] receptionist phone
Thanks Todd.
Can this software work with different brand phones e.g internal has Linksys
and Cisco phone but reception console use Voiceoperatorpanel ?
Thanks,
Sen
I've had good luck with Voice Operator Panel - www.Voiceoperatorpanel.com
Works well, reliable, and quick call processing for large environments.
Their tether feature is the bomb - you can use a good quality phone, with PC
console transacting the calls.
From: sipx-users-boun...@list.sipfou
I can confirm that in 4.2.1 a call to 8+ ext number goes directly to
voicemail.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitchin
(public/usenet)
Sent: Tuesday, July 20, 2010 10:43 AM
To: sipx-users@
lto:tgrazi...@myitdepartment.net]
Sent: Sunday, July 18, 2010 1:17 PM
To: Todd Hodgen
Cc: Josh Patten; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Help with Patton gateway
Sure, but where is it written or set by the project one can no longer do an
attended transfer to a media bas
Analogy Tony. A simple Analogy.
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Sunday, July 18, 2010 12:56 PM
To: Todd Hodgen
Cc: Josh Patten; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Help with Patton gateway
I disagree. Doing an attended transfer to AA
Set expectations. You can't do a supervised transfer to Auto Attendant or
Voicemail in the system today. It's true, it's accurate, it's an
expectation you need to set for the end users. You can dance around this
all you want, but it doesn't change the fact that it isn't supported. Yes,
it sho
There was an in issue with sipxbridge and attended transfers back in 4.0.2 I
believe. There was a patch for it and it was fixed in 4.0.4. Not sure if
this is your issue, I don't see what version you are running on this system.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-bo
Rather than guessing as to what is sending what between these different
devices, it might be prudent to get a PCAP on the server port and look at
the individual packets to see what is being sent out of the server for
caller ID. It might be sending out what you are requesting in the
different flav
Maybe try opening an FTP session and try to download those Config files?
Or, do a PCAP to see what is failing at the server or the phone. You may
need to do a stare and compare between the two of them.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org]
If you use SIP trunks, they will automatically know when the trunks can't
connect to the PBX. You can have your provider set it up to forward on
failure, it would be automagic for you.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ton
If you are using the siptapi program for Outlook click to call, there is an
issue using Outlook 2010 with the siptapi program. However, it's a
Microsoft issue. One simple edit of the registry is required to make it
work. -
http://social.technet.microsoft.com/Forums/en-US/office2010/thread/b85aa3
Having issues with headsets on Polycom 550 phones? A new technical bulletin
says there is an issue with any 550's manufactured between Jan 22, 2010 and
March 31, 2010 and EHS capable headsets - Plantronics, Jabra, etc. Can't be
fixed with software!
http://knowledgebase.polycom.com/kb/search.do?cm
All of my installation are based on ISO. All. I've never seen these issues
you are seeing. Although that might help your situation, it seems it is
because it is replacing something that is broken with a new known working
component.
-Original Message-
From: sipx-users-boun...@list.sipfou
Mike,
Not to be critical here, but this is a key piece of information that hasn't
been in your emails as you explained this dropped call from last night -
that they were on Vonage and could have dropped on their end.
My suggestion here. Forget about all of the past history of this phone did
this
And yet very few people go to the trouble to go to that wiki page and update
the information in it when they find it wrong.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
m...@grounded.net
Sent: Saturday, July 03
An old dog on this list that has gone on to being a full time gamer taught
us this trick about 6 months ago -
logrotate -f /etc/logrotate.d/sipxchange
This will archive your logs and clean up the log directory. Run this.
Then, make a very simple call and run merge-logs from the var/log/sipxpbx
d
There are times when you don't have full access to remote capabilities for
these servers. Having access to port 8443 is one thing, but full access to
their network and the server is sometimes something else. Without it, you
have to go on site to do any of this. If there was a simple option of
is
sipXecs doesn't have it's own soft client if that is what you are asking.
However, it is built on open standards, so most any SIP client for a cell
phone will work for you. Try a google of "cellular sip softclient" or
something similar and you will find pages of clients that claim to be SIP
compli
I don't think enough information has been provided to fully understand what
you are looking for, but I suspect you are looking for a feature like
multi-tenant, which is not support in sipXecs.
You will not keep extensions from calling extensions. You can't have
multiple directories for the email
Was this new approached checked with compliance of the necessary standards.
That's my only thought.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Thursday, July 01, 2010 3:40 PM
To: Tony Grazia
I think the T-1 will give him a better network choice, as you wont have to
deal with the added latency of MPLS. A simple router can provide adequate
QOS for Voice and Data connections, and the cost of a dedicated T-1 from the
Teleco in most areas is very affordable these days, especially for
Gove
Yes, Identical IP/hostname/sipdomain.
Delete and recreate gateway is a good idea I can try.
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Monday, June 28, 2010 8:05 AM
To: thod...@verizon.net; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users
Running sipXecs 4.2.1 (Doug's ISO from Early June) on a server. I've done a
backup of the configuration and the voicemails. Installed from same ISO to
a new server and restored the backups to it. Server is DHCP and DNS for
this network segment.
Issue - can't call out to Gateway. Can ca
Nice find Mike!
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Picher, Michael
Sent: Monday, June 28, 2010 1:52 AM
To: Michael Scheidell; Eric Varsanyi
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sip -> bluetooth
If
The 1535 can do 3 way video conferencing as well. They are a real nuisance
to set up, but I didn't find their quality to be that bad. I used one at my
desk for a few weeks.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On
number, I jut ported the number to ATT and told them how I needed it setup.
-Original Message-----
From: "Todd Hodgen" [thod...@verizon.net]
Date: 06/26/2010 02:20 AM
To: "'M. Ranganathan'" , "'Nathan Nieblas'"
CC: sipx-users@list.sipfound
Broadvox
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of M. Ranganathan
Sent: Friday, June 25, 2010 4:41 PM
To: Nathan Nieblas
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] bandwidth.com problems
O
4.2 was officially released some time ago. It was reported on the
SipFoundry website on Friday, April 16th, 2010.
I'd say use it.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Richard Zhao
Sent: Thursday, June
This mailing list has just become unfriendly.
I agree with Douglas, this is a list about the open source product. If
people are uncivilized on this list, everyone loses.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On B
Is it possible that you have made tweaks to the phones themselves. If they
are like Polycom, once you make a change to the phone, those changes at the
phone override what is put into the group configuration. You can't manage
those changes from the group any longer.
Try defaulting those phones
+1
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of M. Ranganathan
Sent: Wednesday, June 16, 2010 9:40 AM
To: JOLY, ROBERT (ROBERT)
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Tightening up sipxbri
Layne Evans had an answer for this issue back in May, 2009. I've cut and
pasted the majority of his solution below. It's not pretty, but it does
work on the old legacy Polycom phones.
--
The problem is the fact that the polycoms need the whole config from the
sip.cfg and phone1.cfg sent for
I've used that type of configuration in the past, it does work.
Have you tried a transfer to a non-conference (i.e. a regular phone) to see
if that works. That would at least isolate if it is a transfer issue with
your gateway, versus an issue with transferring to the conference bridge.
F
Of the American sipXecs system. Users on this network, in voluntary
co-operation with the SipFoundy have coordinated this test. This is only a
test.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
I have them running on Hardware version 1.0 and firmware 0.1.92
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jim Canfield
Sent: Friday, June 04, 2010 8:16 AM
To: sipx-users@list.sipfoundry.org; sipx-...@sipfoundry.org
Subject: [sipx-u
soft console is not up
and running.
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Thursday, June 03, 2010 6:49 PM
To: WORLEY, Dale R (Dale)
Cc: Todd Hodgen; Hiral Patel; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Time of day routing
You
Are you forwarding it to 8xxx during the off hours? I just tried in on a
system running 4.2.1 and it worked fine.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Hiral Patel
Sent: Thursday, May 27, 2010 9:55 PM
To: sipx-users@list.sipfo
Download Voice Operator Panel at
http://www.voiceoperatorpanel.com/beta/VoiceOperatorPanel-setup.exe
I currently have my CRM integrated with it, in that it does a search based
on CNAM, and pops the record into the integrated browser box.
-Original Message-
From: sipx-users-boun...@list.
users, and
not have to worry about having so many extensions monitoring it.
Or, wait for an answer from someone else as to how many BLF's you can put on
the same park.
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, May 25, 2010 5:45 PM
To: Todd Ho
You have no limit on extensions. You could assign everyone their one Park
position if it's not big of a system.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jim Canfield
Sent: Tuesday, May 25, 2010 5:26 PM
How about trying to export your users list, and then look at that document
for anything that looks out of sorts...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Graeme Allen
Sent: Friday, May 21,
Great news. Thanks!
-Original Message-
From: Mossman, Paul (Paul) [mailto:paulmoss...@avaya.com]
Sent: Friday, May 21, 2010 6:17 PM
To: Todd Hodgen; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] FW: More than one AA dial plan rule? (XX-7822)
Todd wrote:
...
> In my opin
If you have cut and pasted anything, look for trailing characters, including
space. If you have imported any users, also look for characters and spaces
ahead of names, etc.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On B
Where does G711 use 128k of bandwidth.
First, these numbers differ depending on where the call sits in the food
chain. For example, over a T1 connection, it is much less than over an
Ethernet connection. The encoding is different, there is less overhead,
etc. You won't ever see G711 using
ct it should probably be addressed at some
point.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 5/21/2010 1:58 AM, Todd Hodgen wrote:
> Josh, just to be sure you are not missing something here. It is very easy
> to record an Auto Attendant greeting if
, Paul
(Paul)
> Sent: Thursday, May 20, 2010 1:54 PM
> To: Todd Hodgen; 'Alfred Campbell'; sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] More than one AA dial plan rule? (XX-7822)
>
>
>
>>> You can't get rid of this one Paul. Multiple AA&
My apologies, forgot to do a reply all and send to the entire list..
-Original Message-
From: Todd Hodgen [mailto:thod...@verizon.net]
Sent: Thursday, May 20, 2010 11:41 PM
To: 'Mossman, Paul (Paul)'
Subject: RE: [sipx-users] More than one AA dial plan rule
I just entered my conference bridge from an extension, and was able to do an
invite out to a cell phone with no issues. 4.2.1 build.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jim Canfield
Sent: Wednesday, M
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Alfred Campbell
Sent: Wednesday, May 19, 2010 3:54 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] More than one AA dial plan rule? (XX-7822)
On 5/
Yes, on wiki.sipfoundry.org there are several documents.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Wen Jun
Sent: Tuesday, May 18, 2010 9:11 PM
To: sipx-users@list.sipfoundry.org
Cc: 'jun,wen'
Subject: [sipx-
Voip.ms account registered to the same server
account that you have your ITSP sip trunks registered to?
From: Norman Branitsky [mailto:nor...@cherniaksoftware.com]
Sent: Sunday, May 16, 2010 9:13 PM
To: Todd Hodgen
Cc: 'sipx-users'
Subject: Re: [sipx-users] voip.ms registration problem
Under Device, Gateway, your gateway, under ITSP options, click the checkbox
for "register on initialization" box. It is no longer defaulted, and it
should be.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Norma
I am aware of.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Friday, May 14, 2010 6:17 AM
To: sipx-users
Subject: [sipx-users] Fwd: Latest Dev build?
On Fri, May 14, 2010 at 4:56 AM, Todd Hodg
Sent: Saturday, May 15, 2010 1:10 PM
To: Todd Hodgen; sipx-users-boun...@list.sipfoundry.org; 'Tony Graziano';
marce...@discsox.com
Cc: 'D. R. Lang'; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Java goes to 89 percent an web console gets slow
VMware ESXi
CentOS 5.4
Before a question to the dev list, it might be reasonable for those that are
seeing this issue to tell what their configuration consist of. I've seen
one here that is on a VM solution. And Tony is indicating he is running PAE
Kernel.
I recall a message from Scott Lawrence that indicated he didn
issue
here? I suspect it is.
After the reload, it came back with the exact same errors. It's consistent
if nothing else.
Regards,
Todd
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: F
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Friday, May 14, 2010 6:17 AM
To: sipx-users
Subject: [sipx-users] Fwd: Latest Dev build?
On Fri, May 14, 2010 at 4:56 AM, Todd Hodgen wrote
I think a good understanding of echo and where it comes from is helpful.
http://www.ditechnetworks.com/learningCenter/echoBasics.html
It will likely change your perspective of where to search for it.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry
It's a common mistake for people to think that they don't need DNS servers
because they are local to the sipXecs server. DNS is an extremely
important facet of the installation, and the most misunderstood part of the
equation.
DNS is essential to it working. If you configure for an external
What version of Centos is that based on - 5.4 by chance?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Lazieburd
Sent: Wednesday, May 12, 2010 9:57 PM
To: Abdul Mayat
Cc: sipx-users@list.sipfoundry.org
Subject:
Try removing the forwarding that you have set up for the other phones to
ensure its not the problem. When the call goes just to that phone, does it
ring more than once?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behal
x27;known to do this' as this is secondary to phones
working to begin with.
However, I'll follow your input and see if I can gather more info which
might help solve the problem or at least provide input for the next guy to
use.
Mike
On Tue, 11 May 2010 19:26:18 -0700, Todd Hodgen wro
Have you done a trace on these calls to see what is going on?
You might want to do a trace of a call that works, and a call that doesn't.
Get a big table, print it out and do what we call a stare and compare. The
issue should become apparent.
Or, copy your two traces into siptrace and see where
Engineering hat on...
From: Nathaniel Watkins [mailto:nwatk...@garrettcounty.org]
Sent: Tuesday, May 11, 2010 1:59 PM
To: Todd Hodgen; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Linking sipx to another phone system via T1
Todd:
A couple issues/fears with using SIP trunks as
You might consider bringing in SIP trunks for your service, via your
internet connection(s), and a single T1 to your legacy PBX. This will give
you a redundant path for today, and when the Legacy PBX is gone, it can
still be your redundant path, yet you will see the significant cost savings
of usi
http://www.vnet-corp.com/index.htm
This is a second one that works.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence
Sent: Thursday, May 06, 2010 12:40 PM
To: Austin Curry
Cc: sipx-users@list.sipfo
Here is one that I know works with an Asterisk implementation of SIP.
http://pbx2sip.com/?p=56
I have a customer that uses it regularly. Should work fine..
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Austin Curry
Sent: T
Have you tried turning on Nat Traversal? Your original email indicates it
is turned off.
Whether it is the same ISP I don't believe will play into this. At the far
end, are you doing NAT? Are you doing NAT at the near end?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.or
Not sure what changes you are making to the web pages, but there are methods
of making changes via CSS and others that allow those changes to remain and
appear on reboot. It's documented in the wiki. DO a search on sipXecs web
template, that should locate the relevant pages.
-Original Mess
My apology. I'm pulling my foot out of my mouth now
-Original Message-
From: Scott Lawrence [mailto:xmlsc...@gmail.com]
Sent: Friday, April 30, 2010 1:16 PM
To: Andreas (Around the Clock Information Systems)
Cc: 'Todd Hodgen'; 'Scott Ri
You are trying to upgrade from an unstable developer release to a stable
release. I'm sure you will find its not supported, tested, validated or
recommended. You may have other issues to contend with as others have
mentioned, however, the basic process that you are beginning with is flawed.
---
il: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: sipx-users-boun...@list.sipfoundry.org
To: Todd Hodgen ; 'Picher, Michae
That has not been my experience with 4.0.4 and Polycom running 3.1.3c.
Holds and transfers work fine, no problems.
-Original Message-
From: Lara Johnson [mailto:lcr...@ciscorp.biz]
Sent: Thursday, April 29, 2010 5:14 AM
To: Todd Hodgen; 'Picher, Michael'; sipx-users@list.sipf
I can tell you from experience that the behavior of the Polycom 450 will be
significantly better once you get to sipXecs 4.0.2 and above. Your transfer
issues will go away. And you won't have to custom configure the .cfg files.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.
You can see in the log files what the sipxbridge is doing. The log is under
var/log/sipxpbx.
If you can provide the type of phone, version of software, etc. that would
help. Have you tried setting the alias under the attendant to see if it can
answer the calls? At this point, I believe you need
"Use Internal Bridge (SBC)"
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Monday, April 26, 2010 5:39 PM
To: mwh...@thesummit-grp.com; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users]
Of Josh Patten
> Sent: Friday, April 23, 2010 11:33 PM
> To: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] Cisco and sipX 4.2
>
> I KNEW one day I'd see a car analogy on this list :-P
>
> Todd Hodgen wrote:
> > BTW, you can buy a Tom Tom GPS and use
erest in
commercializing it for a virtual server.
From: mkitchin.pub...@gmail.com [mailto:mkitchin.pub...@gmail.com]
Sent: Friday, April 23, 2010 7:32 PM
To: Todd Hodgen; 'Tony Graziano'
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipX 4.2 and VMware?
So is it no
I think you guys are missing a really big point here. A company developed the
support for Polycom so they could sell it in their commercial offering. That
is why it is there, they created it and donated it back to the open source
product, so they would have a product to offer to their customer
What you are calling Cisco bashing is the cold reality that many have found
trying to get a phone developed for a proprietary system to work as a phone
using open standards. We could say the same thing about some of the Nortel
Phones that were designed specifically for a proprietary phone system a
Clouds drop rain on parades..
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, April 23, 2010 1:16 PM
To: mkitchin.pub...@gmail.com
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipX 4.2
Avaya ships the SCS on a DEll Optiplex.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
mkitchin.pub...@gmail.com
Sent: Friday, April 23, 2010 1:10 PM
To: Tony Graziano
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipX 4.
Which development team? The one that created the current interoperability
and donated it, or a new team that is interested in making them work better
and will put together the necessary work to get it done? These seem to be
distinctly different teams than the one that is working on the current
ro
Did you look under User profiles?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
m...@grounded.net
Sent: Thursday, April 22, 2010 8:36 AM
To: sipx-users
Subject: Re: [sipx-users] Custom Notifications
I am still
I can confirm that I had the same issue when I did my update to 4.2
initially. However, I wasn't sure I had a clean system that I was updating,
so I did not report it.
Essentially, everything came up and was working except my trunks. I went in
and added the branch setting - in all areas. I the
If you search the archives, I think you will find a discussion in the past
about the voice used for sipXecs. I seem to recall it about 6 months ago.
They are available, so you should be able to use them for customer messages
that match.
-Original Message-
From: sipx-users-boun...@list.sip
There are currently people using static registrations on sipXecs. I have
one customer with three static registration on three gateways. Works
flawlessly.
However, I have not updated them to 4.2 yet. They are on 4.0.4.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mai
Have you tried it with a soft client to make sure it's not your phones?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael
Scheidell
Sent: Tuesday, April 20, 2010 6:34 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users]
It does a passthrough. If you have a media gateway, it will pass it through
to the end device.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Picher, Michael
Sent: Tuesday, April 20, 2010 4:22 AM
To: Winson (Ela
IF the template is errorring out, I'd recommend documenting it and creating
a JIRA as it doesn't seem to be the desired results.
From: Michael Scheidell [mailto:scheid...@secnap.net]
Sent: Monday, April 19, 2010 12:54 PM
To: Todd Hodgen
Cc: sipx-users@list.sipfoundry.org
Subject:
There was no template for voip.ms when I started using it, so I didn't not
use the drop-down list.
From: Michael Scheidell [mailto:scheid...@secnap.net]
Sent: Monday, April 19, 2010 12:54 PM
To: Todd Hodgen
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] and I am stupid: Re:
And you are making the change both in sipXecs and on their configuration
server?
-Original Message-
From: Michael Scheidell [mailto:scheid...@secnap.net]
Sent: Monday, April 19, 2010 12:40 PM
To: Todd Hodgen; 'Michael Scheidell'; sipx-users@list.sipfoundry.org
Subject: RE: [
You do not need separate sub-accounts for each DID with VOIP.ms. I have 4
on one account, pointing to their Dallas server, and I have one DID for a
foreign exchange in a sub-account that points to their Los Angeles Server.
Works like a champ.
One terminates on the AA, one is pointed to an exte
: Rhon [mailto:c4rdi...@gmail.com]
Sent: Sunday, April 18, 2010 3:28 PM
To: Todd Hodgen; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Managed Phones and Gateways
Hi Todd,
I've seen a lot documentations everywhere (sipxecs.blogspot.com,
myitdepartment.com) are only among th
Al sent an email out last week that listed every change to the software that
comprised the 4.2 release.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Francis Tinio
Sent: Sunday, April 18, 2010 9:43 AM
To: Picher
The wiki is completely built by community involvement. We all have an
obligation to fix what is not accurate, or create articles for the mutual
benefit of the rest of the community.
To quote Scott, "sipXecs is free, but not like beer". A pretty nice
analogy. Our cost as members of this commu
601 - 700 of 963 matches
Mail list logo
