On Mon, May 9, 2011 at 8:40 AM, opyzcinr opyzcinr opyzc...@gmail.com wrote:
What's wrong with FF4?
the phonebook was written in gwt which has all sorts of checks for
browser makes and versions. phonebook app may have to be recompiled
for firefox 4. can you create a bug and reference this
On Mon, May 9, 2011 at 5:50 AM, Kumaran
thiru.venkateshwa...@ttplservices.com wrote:
Please check the scenario which is totally Invalid but to make sipxecs
more friendly and flexible
Scenario:
1.Paging Prefix set to 10 from *77 and Paging number Ext =0
2.When I dial 100 its
On Mon, May 9, 2011 at 1:32 PM, Charles Chalekson chalek...@gmail.com wrote:
Ok, I decided to try to keep my Sonicwall TZ-100 as my DHCP server. I set
option 66 to my pbx server [string type, no other option], option 128 to the
IP address of the server [10.10.10.200], and option 150 with my
On Sun, May 8, 2011 at 9:51 AM, Charles Chalekson chalek...@gmail.com wrote:
One question however, I have always used tftp and know the polycoms out of
the box are defaulted to FTP. Does option 66 also automatically work for FTP
too? I know that sipxecs has a ftp server tuned on by default.
On Sat, May 7, 2011 at 12:18 PM, Charles Chalekson chalek...@gmail.com wrote:
I have had SIPx running for several years without the server running DHCP and
I have just
manually configured phones in the past [long story why I did this but many
tylenol later
I have decided to change].
You
On Fri, May 6, 2011 at 4:30 AM, Kumaran
thiru.venkateshwa...@ttplservices.com wrote:
Please check the below scenario and let me know the update.
1.OpenACD line 650
2.Agent 200 and 201 are logged in and available
3.Caller calls the line 650
4.Agent 200 rings
On Thu, May 5, 2011 at 12:45 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
wasn't there a 4.05 release/patch that you needed to go to to properly
get to 4.2.1?
In any case, why not just erase the offending package/dependency and
then update and let it fix things?
Tony, that was my
2011/5/5 Kemal Eroğlu k.erogl...@gmail.com:
I have a question about DNS and High Availability. I was testing the High
Availability before our product release. I set up a primary and a redundant
server and configured them to share the phone registration load. When all
the phones were registered
On Wed, May 4, 2011 at 9:48 AM, Keith Laidlaw kei...@dakins.ca wrote:
First time I tried it, I got error 404 when I ran “yum update –y” and had to
change “$releasever” to “5” in the base url of the repo then it worked.
you shouldn't have to do this, but i think it's indicative of the
subsequent
On Wed, May 4, 2011 at 9:38 AM, Solomon Mohamed smoha...@yorku.ca wrote:
I changed the STUN server as suggested by Todd to 'stun.ezuce.com'
I don't think Joegen's stun service is up
[dhubler@swift sipxecs]$ telnet stun.ezuce.com 3478
Trying 174.142.82.79...
telnet: connect to address
On Wed, May 4, 2011 at 5:18 PM, Julian Sanchez (alertas)
aler...@bidda.net wrote:
My ITSP is asking me to send SIP 180 Ringing before establish the audio of
the call, this in order to avoid an issue with old PSTN switch were a call
gets answered by AA but after a few seconds gets dropped if
On Mon, May 2, 2011 at 9:22 AM, Anders Mydland and...@mydland.info wrote:
I am testing sipXecs 4.4 with a Norwegian ITSP.
It appears that re-registrations are not sent from sipXecs until a few
milliseconds _after_ the configured register interval.
Consequently, any active calls are
I believe bug fixes like the one George just made to allow conference
pin changes to be made on the fly will get pushed to the 4.4.0 build
sooner rather than later so I put together a release policy open for
comments:
http://wiki.sipfoundry.org/display/sipXecs/Bug+Fix+Release+Policy
This fairly
On Fri, Apr 29, 2011 at 10:10 AM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
Will it be easy/obvious for an user to show exactly what version/fixes
are installed on their system when they are asking for help?
Addressing this and Tony's similar response. This is a good
On Fri, Apr 29, 2011 at 11:52 AM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
I'm far from an expert at yum, repos, etc. Would it be possible for you
to do something link maintain a mirror of the CentOS yum repositiry and
point sipx machines to that repository only? You
On Fri, Apr 29, 2011 at 2:16 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I think it would then be a good idea to be able to display the
versions installed in the gui diagnosticspackage list...
agree so i created
http://track.sipfoundry.org/browse/XX-9575
On Fri, Apr 29, 2011 at 11:39 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
I have one 4.0.4 server that I built from scratch. Postfix is my MTA of
choice, but somehow this machine ended up running Exim.
Exim, really, and sipXecs did that?
I'm fine with
the Sendmail
] migration from ant to automake?
On Thu, Apr 28, 2011 at 12:37 AM, Douglas Hubler dhub...@ezuce.com
wrote:
When trying to add config files for stunnel, I really didn't want to
add to the ant mess in neoconf for installing the new config files so
i converted much of the install target for neoconf
On Thu, Apr 28, 2011 at 2:09 AM, envelopes envelopes
sunwood...@gmail.com wrote:
1st post on the mail-list. I couldn't find the source code repository.
http://wiki.sipfoundry.org/display/sipXecs/Source
branch release-4.2
___
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On Thu, Apr 28, 2011 at 12:10 AM, Jeff Ferrara jferr...@rhino.com.au wrote:
Is there any documentation available regarding how to use the new alert-info
feature? I had a quick look through the wiki and came up only with the 4.4
features list.
Also, can anyone think of a way to use this
On Thu, Apr 28, 2011 at 9:34 AM, Ben Goodfellow b...@btg-computers.co.uk
wrote:
I really like the new internal/external alert feature. However how can I
change the tone generated for external calls? The ring tone set on the phone
affects internal calls, but I cannot figure out how to change
Unfortunately that is bad news. The internal/external ringtone thing is
global setting so one cannot use it if there are any phones configured for
silent ringing.
Is there another polygon setting that can be set to ignore ringtones via
alert info?
On Apr 26, 2011 6:31 PM, Charles Chalekson
On Wed, Apr 27, 2011 at 7:40 AM, Matthew Kitchin (usenet/public)
mkitchin.pub...@gmail.com wrote:
When a phone is set to monitor a parking spot by presence/speed dial, it does
a little chirp when a call goes into the parking spot. I have to set that
line to silent ring. Would it affect that
is this a message reported by your browser? Do you have a screenshot?
On Wed, Apr 27, 2011 at 4:25 AM, Deon Vermeulen
vermeulen.d...@gmail.com wrote:
Hi All
I installed sipxecs 4.4 last night, and everything worked 100%.
I could access the web gui and did the services tests, especially for
On Wed, Apr 27, 2011 at 9:36 AM, Deon Vermeulen
vermeulen.d...@gmail.com wrote:
Attached is a print screen of what I get.
do you use a corporate proxy?
did you modify the internal apache configs at all?
are all the sipxecs running properly? What is returned from command
sipxproc
Did you try
I had the email notifier off by accident. I was missing some great
bug reports against 4.4.0 reported, thanks!
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When trying to add config files for stunnel, I really didn't want to
add to the ant mess in neoconf for installing the new config files so
i converted much of the install target for neoconf/etc to automake and
it's much cleaner IMHO.Would anyone object if i had to convert
more? Maven i know
2011/4/26 Sven Evensen sven.even...@onrelay.com:
We are trying out the Pidgin client for corporate IM. We also want to pick
up phone presence, at least
the users who have proper desk phones. But if a user logs off their IM
client (thus showing status Offline),
the “On the phone” presence
On Tue, Apr 26, 2011 at 12:27 AM, Charles Chalekson chalek...@gmail.com wrote:
Unsure how this can be related to 4.4 but I just upgraded, and have several
polycom 650s. Prior to upgrade to 4.4, I had several phone lines on my
polycoms set to silent ring [no audio ring, flashing light only].
I will try to restore them from few weeks back. I'll keep you posted.
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On Mon, Apr 25, 2011 at 2:19 PM, George Niculae geo...@ezuce.com wrote:
On Mon, Apr 25, 2011 at 9:10 PM, Burleigh, Matt
matt.burle...@eiisolutions.net wrote:
Understood, but if a user changes his PIN in the middle of the night,
they’ll have to wait until I notice that the restart needs to
track/wiki/www.sipfoundry.org are back online. www is missing a few
images, but I've let the right people know about that. let me know if
you have any issues. sorry for the outage.
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List
it's down and does not look good. system is wedged, won't even reboot
properly. it may take a while to get it back online, i'll keep folks
posted
On Thu, Apr 21, 2011 at 8:48 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Was trying to test an update to 4.4. Can't reach the site.
--
On Thu, Apr 21, 2011 at 1:17 PM, Josh M. Patten jpat...@co.brazos.tx.us wrote:
Perhaps this is why:
http://slashdot.org/story/11/04/21/1515238/Major-Outage-At-the-Amazon-Web-Services
This is exactly why. It started w/freezing when accessing the EBS
volumes and now i cannot even take a
It actually never went down
http://download.sipfoundry.org/pub/sipXecs/
It is on a different host.
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On Thu, Apr 21, 2011 at 10:01 PM, Outback Dingo outbackdi...@gmail.com wrote:
isnt there an install iso someplace ?
http://download.sipfoundry.org/pub/sipXecs/ISO/
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On Wed, Apr 20, 2011 at 7:59 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
to what end? openfire sip plugin will register, but it cannot transfer
calls or make insternal calls either (does not support refer)
Good question, Darth, what is your desired goal?
On Wed, Apr 20, 2011 at
On Tue, Apr 19, 2011 at 4:37 PM, Jim Canfield jcanfi...@emstar.com wrote:
Still not sure how the endpoint settings should be setup though, when
I go available my phone rings, but i cannot answer, it just gets
stuck in limboland. After it's stuck, I see the agent in queue with
timer running
Finally! Thanks to everyone for making this happen
http://download.sipfoundry.org/pub/sipXecs/
Instructions
http://wiki.sipfoundry.org/display/sipXecs/Upgrade+to+Latest+Stable+Version
Going forward, there will be updates to this build for bug fixes as
they are fixed. I will publish a
On Wed, Apr 20, 2011 at 7:22 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
I thought those screen shots were from sipXecs…now I realize those are
openACD screens J
However, all is not lost. You can safely edit queues, recipes, etc. on
the admin interface on port that comes with
If you make someone a supervisor, login to interface on port 5050 then add
supervisor dashboard you should see all sorts of info.
The format for the commands is quasi freeswitch but on your initial tests,
ignore those commands and log into interface on port 5050 and go available
from there to
On Tue, Apr 19, 2011 at 3:37 PM, Michael Scheidell
michael.scheid...@secnap.com wrote:
On 4/19/11 3:20 PM, Douglas Hubler wrote:
If you make someone a supervisor, login to interface on port 5050 then add
supervisor dashboard you should see all sorts of info.
sorry, this is not clear at all
On Sat, Apr 16, 2011 at 11:59 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Fyi - last ISO 4.4 does not run sipxconfig web interface...
32 or 64 bit? just tried 64 bit ISO and didn't have any issues w/ui
coming up.
I'm rebuilding now because we want to get the latest fixes in.
On Fri, Apr 15, 2011 at 11:42 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Seeing a lot of dependency issues... I also don't think this is hard
to overcome for some users, but wanted to share it.
***
-- Finished Dependency Resolution
sipxecs-4.2.1-18591.9.1.i386 from installed has
On Fri, Apr 15, 2011 at 3:15 AM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
Is anyone having trouble using Eric Josh's Custom polycom config -
http://sites.google.com/site/sipxecstipsandtricks/polycom-phones
This is how 4.4 implemented the custom configs
Ugh, this should have been made available from the UI. Sven,
hopefully you can get by, by editing your customer configs?
I created this...
http://track.sipfoundry.org/browse/XX-9562
and selected for 4.6
On Fri, Apr 15, 2011 at 10:54 AM, Sven Evensen sven.even...@onrelay.com wrote:
Media
No binaries published for ubuntu. Best luck is googling for compiling from
source
On Apr 16, 2011 2:46 PM, Aza Tek azat...@gmail.com wrote:
Is there a stable repo for sipxecs 4? I'm using Ubuntu 10.10.
Thanks for sipXecs...
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If all goes well. Lingering issues that need to be cleaned-up, but
this is mostly a waiting period for any new issues at last chance
fixes to come in.
I refactored the upgrade instructions extensively to make upgrade
process straightforward as possible
On Fri, Apr 8, 2011 at 4:35 PM, Jim Canfield jcanfi...@emstar.com wrote:
Can anyone confirm that the Openfire admin access is broken on 4.2.1
http://wiki.sipfoundry.org/display/sipXecs/The+Openfire+admin+console
This procedure no longer seems to work.
do you have any details on what didn't
FYI: Just wanted to let folks know, I'm listening even if i don't have
the time to address each issue.
As you can imagine, sparse documentation makes things very difficult
for ezuce engineers as well so we feel your pain too. Good news is
that now that the wiki is organized, ezuce is starting to
There are 1000+ issues/feature requests/improvements/tasks in jira in
various states of being valid. They are also are not in any order of
importance. I'm going to be cleaning up jira and defining a process
that will make jira more useful as a tool for us. I have a pretty good
idea of what needs
On Fri, Mar 25, 2011 at 11:30 AM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
The wireshark shows it pretty clear to me, if Im reading it correctly. It
is attached, and I will attach it to the jira to.
Call is from 6155008073 to 6159253938. It should forward to
On Tue, Mar 29, 2011 at 3:16 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
Sure. Merged logs are attached. This is a call from 6159253938 (sipx 4.4
test) to 6154670142 (sipx 4.2.1 production). The 2 systems are on seperate
Verizon sip trunks.
Would you like to see a
On Tue, Mar 29, 2011 at 3:38 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Tue, Mar 29, 2011 at 3:16 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
Sure. Merged logs are attached. This is a call from 6159253938 (sipx 4.4
test) to 6154670142 (sipx 4.2.1 production). The 2
On Tue, Mar 29, 2011 at 4:35 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Tue, Mar 29, 2011 at 3:38 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Tue, Mar 29, 2011 at 3:16 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
Sure. Merged logs are attached. This is a call
On Tue, Mar 29, 2011 at 5:36 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
On 3/29/2011 4:11 PM, Michael Scheidell wrote:
On 3/29/11 4:41 PM, Douglas Hubler wrote:
Obvious answer is that there are no credentials, sipxecs is never
challenged by verizon. In addition
From ITSP Account: X
On Tue, Mar 29, 2011 at 8:29 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
On 3/29/2011 6:55 PM, Douglas Hubler wrote:
On Tue, Mar 29, 2011 at 5:36 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
On 3/29/2011 4:11 PM
On Tue, Mar 29, 2011 at 9:05 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
Is it significant that it is showing different numbers? 4.2.1 is showing the
cell phone I'm calling in on. 4.4 is showing the DID alias for the extension
that is being forwarded.
As for the ITSP
On Tue, Mar 29, 2011 at 9:45 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
The notes next to that setting indicate:
Set the From header to ITSP account information for outgoing INVITE
messages. If you need to set this, ITSP is not following RFC and should be
notified to
Questions:
Can explain the simplest scenario that fails?
Explain exactly what external calls is? incoming, outgoing or both
Does the scenario start with a call between a polycom hardphone and
spectralink registered as two different extensions?
On Mon, Mar 21, 2011 at 12:39 PM, Matthew Kitchin
On Tue, Mar 22, 2011 at 4:44 PM, Matt White mwh...@thesummit-grp.com wrote:
On 3/22/2011 at 04:19 PM, in message 4d890466.6060...@secnap.com, Michael
Scheidell michael.scheid...@secnap.com wrote:
where are you specifying an stun server? what version of sipx is this?
I didn't think sipx
Fixed, ISO page will fix itself on next ISO update.
On Fri, Mar 18, 2011 at 2:26 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
The download.sipfoundry.org site has a link on all the pages that evidently
became stale when the wiki was reworked.
The sipXecs Start Here link points to:
On Wed, Mar 16, 2011 at 5:02 AM, Irena Dolovčak
irena.dolov...@gmail.com wrote:
Thanks Douglas,
it's working now.. but still not as I want to..
the number is shown when I put it in the preferred identity like this:
385...@sipconnect.sipgate.de
is there a way to make every user to use a
On Wed, Mar 9, 2011 at 4:12 PM, Michael Scheidell
michael.scheid...@secnap.com wrote:
so, how do I put it on a submenu, and is cookbooks/alerts and notifications
best place?
yes, i think so
2 ways:
1.) Edit page and you will see Change location toward bottom of edit page
2.) Tools/View
On Tue, Mar 8, 2011 at 3:12 AM, Irena Dolovčak irena.dolov...@gmail.com wrote:
Great!
Thank you very much Douglas.
Can you tell me the exact version where this is fixed? (just to be sure, is
it the version sipxecs-4.4.0-129.ga7ccc-x86_64.iso ?)
It's building now actually, there was a build
Just to keep people posted, there is a SF board meeting this Thursday
where this will try to be decided and as it stands now, I will have a
strong vote against this as I outlined. I could change my vote should
new information emerge. If you have a response on either side that
you haven't
On Sat, Mar 5, 2011 at 5:54 AM, Irena Dolovčak irena.dolov...@gmail.com wrote:
Hi Douglas,
here is the error that i get when the trunk fails on 4.2.1
in the Prefered identity field i set this: 385492...@sipconnect.sipgate.de
and get this error:
Let's focus on 4.4 if we can, you said both
On Mon, Mar 7, 2011 at 4:04 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Sat, Mar 5, 2011 at 5:54 AM, Irena Dolovčak irena.dolov...@gmail.com
wrote:
Hi Douglas,
here is the error that i get when the trunk fails on 4.2.1
in the Prefered identity field i set this: 385492
On Mon, Mar 7, 2011 at 4:31 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Mon, Mar 7, 2011 at 4:04 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Sat, Mar 5, 2011 at 5:54 AM, Irena Dolovčak irena.dolov...@gmail.com
wrote:
Hi Douglas,
here is the error that i get when the trunk fails
It doesn't seem as though you addressed the folks writing plugins in a
non-hosted environment and that sell packaged solutions. This change
would affect you as a developer/company in the following way:
As a developer, you would have to make the source available to your
plugin the instant there
On Fri, Mar 4, 2011 at 10:47 AM, Michael Picher mpic...@gmail.com wrote:
Jeff,
When we find bugs or would like a feature enhancement the best place to file
something is in the Jira (http://track.sipfoundry.org).
This sounds like a logical enhancement / change.
Also, I'm not aware of any
4.4.0 trace only shows a registration, not an INVITE attempt thru the trunk
On Fri, Mar 4, 2011 at 5:46 AM, Irena Dolovčak irena.dolov...@gmail.com wrote:
here are the logs from both servers.
By crash i mean that the configuration test fails. (system - services -
SIP trunking)
ok, this is a
On Thu, Mar 3, 2011 at 8:38 AM, Irena Dolovčak irena.dolov...@gmail.com wrote:
Hi,
I'm using sipx 4.2.1 with SipGate sip trunk and i can't make the trunk to
show the caller ID number. In some scenarios it does work thou.
I managed to make the number to show on mobile phones, but when I call
On Thu, Mar 3, 2011 at 8:38 AM, Irena Dolovčak irena.dolov...@gmail.com wrote:
the funny thing: when I try to put the preferred identity, whatever I set in
that field, the SIP trunking service crashes. (same thing on 4.4.0)
what do you mean by crash. You have the message?
can you capture
On Wed, Mar 2, 2011 at 9:36 AM, Geoff Van Brunt gvanbr...@dstgroup.com wrote:
The problem is the domain is included in calculating the hash. So if you
restore to a new domain it uses the new domain for calculations to compare
against so they don't match. What is needed is to drop the domain
On Wed, Mar 2, 2011 at 8:02 PM, Michael Scheidell
michael.scheid...@secnap.com wrote:
using iso from yesterday, goal is to try to get a working 4.4.0 system as a
'cold standby', and if it works, upgrade my 4.2.0 to it.
(hopefully to get voip.ms to do ip based registration on port 5060!)
so,
On Wed, Mar 2, 2011 at 8:59 PM, Michael Scheidell
michael.scheid...@secnap.com wrote:
i seem to remember that 4.2.1? needed more 'hops' than 4.2.0
Can you capture the incoming SIP messages?
Specifically, is Max-Forwards specified in the header? How long is
the Via list?
On Tue, Mar 1, 2011 at 11:27 AM, m...@grounded.net m...@grounded.net wrote:
On Tue, 1 Mar 2011 11:16:18 -0500, Tony Graziano wrote:
as always, there is, you just arent looking in the ISO directory
http://download.sipfoundry.org/pub/sipXecs/
http://download.sipfoundry.org/pub/sipXecs/ISO/
On Tue, Mar 1, 2011 at 1:24 PM, Michael Scheidell
michael.scheid...@secnap.com wrote:
On 3/1/11 1:17 PM, Douglas Hubler wrote:
You caught me in the middle of updating the ISOs. Refresh page in 5 min.
what is the memory leak fix?
memory leak was for 4.0.4 systems in HA mode. fix was already
On Tue, Mar 1, 2011 at 1:35 PM, Michael Scheidell
michael.scheid...@secnap.com wrote:
rumor has it, you can use port 5060 for both trunking and remote users/sip:
calls.
If I set it up this way, is the src port on sipx also 5060? I guess it
would have to be for bydirectional 5060/5060
On Tue, Mar 1, 2011 at 12:41 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Right. Manual iso install, restore users/phones via import. Manually add
gateways, etc.
My pledge
We will not ship another sipXecs with solving back/restore to
separate domain or ip again.
I hit this all the
On Tue, Mar 1, 2011 at 3:29 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Agreed, it's a royal pain. Does the JIRA have a pledge request (so we can
track that)? LOL.
There are probably several ways to skin this, but I think it's been
discussed before. The single biggest obstacle being
On Mon, Feb 28, 2011 at 12:00 PM, m...@grounded.net m...@grounded.net wrote:
On Mon, 28 Feb 2011 16:27:49 +, Josh M. Patten wrote:
Certainly not until http://track.sipfoundry.org/browse/XX-9475 is fixed.
Darn, guess there's no stable version for those running 61 users or less eh?
:)
y,
4.4 is out of dev. stage so you will be able to get updates to it,
however that doesn't mean a whole lot because i still wouldn't
recommend you upgrade a production system to it yet.By test, i
mean setup a spare box and verify the features you use most still
work.
On Mon, Feb 28, 2011 at
In my copious spare time, I'm reinstalling the entire atlassian stack
with the latest versions of everything. Main driver is that we've
found someone that has figured out how to integrate liferay forums
w/atlassian stack but needs a do-over with how everything is setup.
I'm hoping this will
On Wed, Feb 23, 2011 at 8:59 AM, Michael Scheidell
michael.scheid...@secnap.com wrote:
I have a remote sipx server I am trying to configure as an ha server.
previously, I had restored our main sipx config to it (so it thinks its the
primary server), but to fix it, I tried to run sipxecs-setup
On Wed, Feb 23, 2011 at 2:56 PM, Michael Scheidell
michael.scheid...@secnap.com wrote:
I am sure I. Really borked it up. Esp if I still had a web interface on
secondary.
I will try a custom config. If all else fails. Ill just put up fs to do aa vm
and forward
I you go this route, FS that
On Mon, Feb 21, 2011 at 9:14 PM, Alistair J. Fenning
alist...@afonweb.co.uk wrote:
As far as I can see it is not currently possible to have sipXecs try and
match incoming caller ID with a name (either from the inbuilt Phonebook or
external service) and then re-write the SIP message with the
On Mon, Feb 21, 2011 at 5:45 AM, George Niculae geo...@ezuce.com wrote:
Nice patch - have a question here, should the feature within
generated *-sip.cfg.xml contain the newly introduced custom-configs?
Right now it appends it as an attribute, (notice
feature.19.name=custom-configs and
On Sat, Feb 19, 2011 at 11:31 AM, Douglas Hubler dhub...@ezuce.com wrote:
On Fri, Feb 18, 2011 at 11:36 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Fri, Feb 18, 2011 at 10:58 PM, Jim Canfield jcanfi...@emstar.com wrote:
On Fri, Feb 18, 2011 at 2:46 PM, Josh M. Patten jpat...@co.brazos.tx.us
On Fri, Feb 18, 2011 at 11:36 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Fri, Feb 18, 2011 at 10:58 PM, Jim Canfield jcanfi...@emstar.com wrote:
On Fri, Feb 18, 2011 at 2:46 PM, Josh M. Patten jpat...@co.brazos.tx.us
wrote:
Hey if anyone uses the custom config files patch written by Eric
On Fri, Feb 18, 2011 at 10:58 PM, Jim Canfield jcanfi...@emstar.com wrote:
On Fri, Feb 18, 2011 at 2:46 PM, Josh M. Patten jpat...@co.brazos.tx.us
wrote:
Hey if anyone uses the custom config files patch written by Eric Varsanyi
that I’ve been posting about lately and would like it to appear
On Thu, Feb 17, 2011 at 10:52 AM, andrewpitman andrewpit...@comcast.net wrote:
Content-Type: text/plain;
charset=utf-8
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Organization: SipXecs Forum
In-Reply-To: de56.4d5bd...@forum.sipfoundry.org
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 56971
Message-ID:
On Wed, Feb 16, 2011 at 10:13 AM, Bob Blanchard Jr. bl...@dainty.ca wrote:
I have found a hack to change the language on a transfer to another
auto-attendant, involving passing locale on a transfer to a URL.
First, create the second-language auto-attendant (in our case french) and
determine
no issues filling.
it should at least be configuable
On Wed, Feb 16, 2011 at 12:26 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
I'm not sure if this is true in all versions, but in 4.4, you cannot restore
a backup over about 1 GB using the upload feature in the GUI.
On Tue, Feb 15, 2011 at 12:14 PM, Victor Williams
vbwilliams.l...@gmail.com wrote:
Only thing I would change is the font. Comic Sans looks a bit...comicky. I
prefer Arial...easier to read and looks a bit more professional.
Mock-ups are meant to be more like pencil sketches, common technique
On Tue, Feb 15, 2011 at 1:30 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
I am up and running with a Polycom Spectralink 8020.
I had tried this model once before but gave up.
There were 2 big differences this time.
1) I called Polycom sales first and was given the
The ezuce team: joegen, cristi, mircea, george, laurentui and myself
have just completed a major update to the wiki documentation over 2
full days. We tried to remove old content, update existing content
and create critically missing content. We also spent a lot of time
organizing the content
On Mon, Feb 14, 2011 at 12:51 PM, Geoff Van Brunt
gvanbr...@dstgroup.com wrote:
One thing absent which would be awesome is the re addition of the road
map... :)
Very true!
Next up is jira cleanup and that will help drive roadmap and release notes.
On Mon, Feb 14, 2011 at 2:31 PM, Michal Bielicki
michal.bieli...@seventhsignal.de wrote:
Isn't there a function in the wiki to read the stuff out of jira ?
yeah, i plan to take advantage of this once jira is cleaned up.
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