On Fri, Jul 2, 2010 at 7:41 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
3. Have the merged.xml ready for download in sipxconfig (nice, comments
about what it is, storing it there for future reference) to display in
viewer.
it might be possible to wrap the sip viewer as an applet so
On Fri, Jul 2, 2010 at 9:09 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Smells like an applet, but I don't see how that would make it easier.
applet could build a list of calls, or servers, or phones, can you can
pick them from a drop down and applet would show just that data
On Wed, Jun 30, 2010 at 2:58 PM, m...@grounded.net m...@grounded.net wrote:
I'm beat, no idea how to update short of installing 4.2.1 using ISO and
restoring from a backup.
Thank you all filling mike in, I was not paying attention. You all
pointed him in the correct direction
As far as what
On Tue, Jun 29, 2010 at 12:45 PM, Gerald Harper ger...@sustaa.com wrote:
Thanks Sven, I may try this on a 4.2 box this weekend. Does anybody have any
hints, tips or suggestions?
if you run into trouble, or looking for an alternative, you may try
looking into these guy's FS mod
On Fri, Jun 25, 2010 at 1:22 PM, JOLY, ROBERT (ROBERT) rj...@avaya.com wrote:
The bug introduced by 18076 got undone in revision 18944 so presumably hunt
group call pick up started working again after that revision...
Robert,
This checkin seems to exclusively deal with the addition of the
On Tue, Jun 29, 2010 at 12:14 AM, Josh Patten jpat...@co.brazos.tx.us wrote:
*bump*
On 06/26/2010 12:49 PM, Josh Patten wrote:
*bump*
you're not afraid of the double-bump ;)
On 06/24/2010 05:39 PM, Josh Patten wrote:
OK I think I have figured out why this is happening and I think it's a
On Sun, Jun 27, 2010 at 9:47 AM, mattias jonsson m...@mjw.se wrote:
Wich ubuntu will sipbx work with?
there are no binaries. the closest I know of is if you search the
list, you'll find experiences compiling the source on debian. if you
have patches, I'll happy review and accept them, i would
On Wed, Jun 23, 2010 at 7:23 AM, Staffan Kerker ietf-li...@kerker.se wrote:
Now my question is, am I missing something here that will break stuff?
host1 (configuration only)
--
Configuration/Management
Primary SIP Router (shutdown)
host2 (the new primary)
--
Voicemail
i thought the
On Wed, Jun 23, 2010 at 7:50 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
http://en.wikipedia.org/wiki/List_of_SIP_software
kapanga is not on that list and my friend in the biz thinks it's
pretty good. windows, android, windows mobile
http://www.kapanga.net/
On Wed, Jun 23, 2010 at 9:46 AM, Picher, Michael
mpic...@cmctechgroup.com wrote:
It looks like Freeswitch could do this with mod_xml_curl …
I’m just not into programming pain. :-)
What would be the pain point? getting the module available or working
with the module? if it's getting the module
On Tue, Jun 22, 2010 at 10:13 AM, Michael Scheidell
scheid...@secnap.net wrote:
sipx doesn't need, doesn't use the cisco dialplan.xml.
cisco phones were able to dial 4 digit extensions, as well as outdial
without hitting the '#' or dial button or waiting the 5 second timeout.
On the 7960/40
On Tue, Jun 22, 2010 at 10:51 AM, Matt White mwh...@thesummit-grp.com wrote:
Let us know if you actual are able to buy one. As that seems to be the
issue.
There is already enough confusion about SCS v.s. sipXecs on this
mailing list. If someone creates a mailing list or google group to
talk
On Tue, Jun 22, 2010 at 11:49 AM, Gerald Harper ger...@sustaa.com wrote:
Not to be rude, but who the hell made you God? This is supposed to be a
discussion of all things sipXecs and I had a question regarding SCS vs
sipXecs, not my fault the thread was hijacked to be a does it exist
question.
On Tue, Jun 22, 2010 at 10:00 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
I need to interface with sipXecs via SOAP to add users via another web page
I'm not too familiar with where the REST api has gone, i don't see
user crud calls
On Mon, Jun 21, 2010 at 5:22 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I did an update about 4 weeks ago to a 3.10.3 system, and brought it
up to 4.0.4. I've noticed there are no CDR's in the database prior to
the upgrade time.
Did you mean prior or post?
On Mon, Jun 14, 2010 at 12:31 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
The notes implied it had been merged to main...guess I'll go yum and see what
happens.
it definitely went to 4.2 but was not built by avaya yet AFAICT.
I have ISOs and RPMs that are up-to date here
On Mon, Jun 14, 2010 at 2:47 PM, Josh Patten jpat...@co.brazos.tx.us wrote:
You'll also notice some extra features in that build (assuming it's the
OpenSuSE build service one) ;-)
To get the list of fixes, look for issues submitted by lazyboy that
are in patch pending state. It also includes
On Mon, Jun 14, 2010 at 4:21 PM, Tran, Ly V. lt...@rrtgi.com wrote:
Just found the thread in dev,
http://forum.sipfoundry.org/index.php?t=msggoto=47588S=85598d4474612ad
bb2100dff584a5de0
Need to do some more reading, but looks interesting.
It will probably be a new module eventually, but it
On Thu, Jun 10, 2010 at 4:54 AM, Massimo Vignone
massimo.vign...@unimore.it wrote:
After upgraded to 4.2 release, i can't restart Linksys and Polycom
phones via GUI.
you can use ngrep, tcpdump or wireshark to watch the NOTIFY message
coming from sipXconfig on 5060 to compare it with the one you
On Wed, Jun 9, 2010 at 10:12 AM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
The gmane.comp.telephony.pbx.sipfoundry.general rss feed does't seem to be
working. I'm not sure how used this is - just thought I'd mention it.
rss i recommend it to folks looking to watch what's going on.
On Wed, Jun 9, 2010 at 12:55 PM, WORLEY, Dale R (Dale)
dwor...@avaya.com wrote:
The gmane.comp.telephony.pbx.sipfoundry.general rss feed does't seem to be
working. I'm not sure how used this is - just thought I'd mention it.
___
Isn't that Gmane's
On Tue, Jun 8, 2010 at 12:07 PM, Abdul Mayat abdul_ma...@hotmail.com wrote:
I thought it would be useful to also include email
notification method and External MWI as additional fields
(these would allow a full voicemail deployment to be
uploaded).
Using the REST API might be a better place
On Tue, May 25, 2010 at 12:34 AM, Josh Patten jpat...@co.brazos.tx.us wrote:
4.28.1
The one included with the community build using the openSuSE build service.
I'm not entirely sure why 4.28.1 was chosen.
The stunnel that comes w/CentOS was apparently not sufficient for
whoever added the
On Tue, May 25, 2010 at 10:56 AM, Josh Patten jpat...@co.brazos.tx.us wrote:
All that would be required to make the new version work would be to put fips
= no in the autogenerated stunnel config on the primary server
I see, I've had added the fips=no in the latest build about a week ago
On Tue, May 25, 2010 at 11:09 AM, Mossman, Paul (Paul)
paulmoss...@avaya.com wrote:
But I suspect that real world users are bothered more by the vice-versa Line
case: Adding a Phone Line to a selected User.
I guess it depends if you normally add users to phones or phones to users.
For adding
On Tue, May 25, 2010 at 1:25 PM, Mossman, Paul (Paul)
paulmoss...@avaya.com wrote:
It would be nice if the table contents updated in the backgroun as you type.
i.e. Without the need for a screen refresh.
you can drop the entire search page and add one ajax text field to
lines page. It would
On Tue, May 25, 2010 at 2:02 PM, Mossman, Paul (Paul)
paulmoss...@avaya.com wrote:
Douglas wrote:
I can put together a screen mock-up if you want.
Sure.
I attached screenshots
http://track.sipfoundry.org/browse/XX-8462
And Dale, if autocomplete doesn't kick in, then, yes, if you simply
type
On Mon, May 24, 2010 at 4:39 PM, IT Services itservi...@apiwellness.org wrote:
As the default, the polycom phone display shows the extension number. I
want to display the first name of the user.
I would think you can change the screen name field on the SIP Line
Settings. For example, if the
On Mon, May 17, 2010 at 4:47 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Installing fresh from this iso with manual format does not allow partition
options and formats and installs like the automatic format.
i fixed this part and rebuild ISO and will test in AM.
Sorry i've been slow
On Sat, May 15, 2010 at 2:23 AM, Todd Hodgen thod...@verizon.net wrote:
Doug, I did a re-install from the ISO and confirmed that when I select the
installation and the format option, the system installs but it never stops
at the format screen to request the type of format. Is that something
On Fri, May 14, 2010 at 4:56 AM, Todd Hodgen thod...@verizon.net wrote:
During the bootup process there was the following error - Starting
Phonelogd: Configuration file not found: /etc/sipxpbx/phonelog-config
(Unconfigured)
i'll take a look. If the dependencies are not right, random ordering
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