On Fri, Mar 23, 2012 at 3:01 PM, Simon Brûlé sbr...@360-innovations.com wrote:
I try to put the auto-attendant in french and english by redirecting the
main voice menu to an other dial plan by giving him the option LOCALE= and
I also tryed with a phantom user and it doesn't work either.
check
On Tue, Mar 20, 2012 at 11:29 AM, Михаил Родионов ma...@siplabs.ru wrote:
My 5 cents...
I think we are not talking xhtml here but we are talking that having some
place to put custom files and/or scripts to be used by phones (for example,
xml-producing scripts, etc) inside the apache web root
Nettica.com should work.
If you only have one server, you don't need srv records, but dns should be
running on sipx server and it should be configured automatically
On Mar 19, 2012 3:53 PM, Stiles Watson wat...@datatek-net.com wrote:
Are there instructions for setting up SRV records with
On Thu, Mar 15, 2012 at 4:31 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
P.P.S. We using the apache server now that's managed by the OS
instead of launching an apache instance with a separate config file.
YAY! So this config change goes in the sipx config file. do you mean it will
On Thu, Mar 15, 2012 at 5:22 PM, Gerald Drouillard
gerryl...@drouillard.ca wrote:
Try this:
Location /sipxconfig
RewriteEngine on
RewriteCond %{HTTPS} !=on
RewriteRule ^.*$ https://%{HTTP_HOST}%{REQUEST_URI} [L,R]
#if proxy to a different box using https enable the next 2 lines
#SetEnv
On Thu, Mar 15, 2012 at 4:37 PM, Aaron Pursell aar...@esgw.org wrote:
Interesting and good to know while I work on my 4.6 lab to get ready for it.
Only problem is I have been running the native HTTPD on port 81 to allow for
a weather page we display on all our Polycoms idle screens, so will
Are you update 4.4 and have you updated to the latest build. There
were fixes here according to Mircea
On Fri, Mar 16, 2012 at 11:42 AM, Clayton Gooding
clayton.good...@gmail.com wrote:
Hello
We have a deployment of SipX where we require an SSL certificate to prevent
IE users from getting
On Fri, Mar 16, 2012 at 12:48 PM, Clayton Gooding
clayton.good...@gmail.com wrote:
Thanks Douglas, it looks like that link was authored about a month ago which
I would think indicates we should have captured the correction during the
yum update about 1 month ago. But I am still rather green
Sorry, I meant this for list.
Kumaran, FYI: no need to CC developers, we all get emails from mailing list.
Also, logs were helpful, I should have fix for that page error in next build.
-- Forwarded message --
From: Douglas Hubler dhub...@ezuce.com
Date: Thu, Mar 15, 2012 at 9:04
I had to unraveled apache config in 4.6 when we introduced cfengine.
For admin interface I decided to go for simple apache config instead
of static html landing page w/meta tag. I came up with an apache
config. This redirects http requests to admin ui to https and
redirects / to /sipxconfig.
On Tue, Mar 13, 2012 at 2:04 AM, Emilio Panighetti emilio...@me.com wrote:
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To: 10312.4f4e6...@forum.sipfoundry.org
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 66549
Message-ID:
On Tue, Mar 13, 2012 at 5:17 AM, Emilio Panighetti emilio...@me.com wrote:
That's interesting. So if I read you correctly; sipXecs is a
stateless proxy only? Similar in objective to SER and its
derivative projects?
sipXecs includes a stateless proxy called sipXproxy (like SER) and
would be the
On Sun, Mar 11, 2012 at 1:43 PM, Robert B d...@spudland.com wrote:
I've been asked to make a change to the auto-attendant if possible...
I've made a test change to
/usr/share/www/doc/aa_vxml/autoattendant-operator.vxml (I removed the
+ 'please_hold.wav' to the extension transfer section.
I
On Fri, Mar 9, 2012 at 1:43 PM, Joe Micciche jmicc...@redhat.com wrote:
1 - configurable, for example I have the option to (dis)allow mp3;
yes
2 - extended to enable other formats, e.g. ogg
Much of the work dealt with freeswitch config, so I'm sure this will
be possible.
Don't worry, I can safely say MP3 would *never* be the only option,
license issues or not.
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On Thu, Mar 8, 2012 at 1:15 AM, Todd Hodgen thod...@frontier.com wrote:
Douglas – will there be any published results for Hackfest?
A lot of the guys are visiting US for a while, but I can give you some
info i gathered. This is all work that will go into 4.6
Da Gangstas!
* We've
A lot of the guys are visiting US for a while, but I can give you some
info i gathered. This is all work that will go into 4.6. Maybe
they'll respond with more info.
Da Gangstas!
* We've worked thru the homer integration steps and don't expect any
surprises capturing that in rpm
On Thu, Mar 8, 2012 at 11:39 AM, Michael Picher mpic...@ezuce.com wrote:
try re-running the whole setup i guess with one nic. this will wipe your
system!
Have to agree here, error in log you sent wasn't conclusive of anything.
The ACD in sipXecs works for many people, but definitely has had
On Thu, Mar 8, 2012 at 1:03 PM, Douglas Hubler dhub...@ezuce.com wrote:
Night Losers
* Option to save voicemail saving to MP3 working.
Nathaniel Watkins wrote
Douglas - can you clarify - did MP3 voicemail encoding make the cut?! If
not, I demand a re-vote - or a revolt...I'm
On Fri, Mar 2, 2012 at 4:59 AM, Borginger Rikard
rikard.borgin...@combitech.se wrote:
Would it be better to ask this question to the developer-list instead?
this list is fine, every developer also subscribes here and this is
more of a user question
Mike sent the list, but beyond that, are there
On Thu, Mar 1, 2012 at 6:09 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I thought too that had been fixes. What version are you running?
UTF or PCMU?
I could be wrong, but I don't recall either being fixed, but I recall
PCMU restriction discussed as restrictive.
On Thu, Mar 1, 2012 at 6:52 AM, George Niculae geo...@ezuce.com wrote:
This is the topic on PCMU:
http://forum.sipfoundry.org/index.php?t=treeth=16017S=30249bec0702ee12c59157848a823af7
What was final result of PCMU restriction, is it in 4.4?
___
On Sat, Feb 25, 2012 at 12:56 PM, Becker, Jesse beck...@sunyulster.edu wrote:
All,
Is there a file that can be modified to allow custom ring tones to be
available in all phone profiles? Currently, we have been setting custom ring
tones under sample audio using the web interface of the phone
On Wed, Feb 22, 2012 at 1:30 PM, Becker, Jesse beck...@sunyulster.eduwrote:
We used My Buddy pretty heavily ( 3 ) in our OIT office. However,
sometimes certain components stops functioning. Most noticeably the find
feature, call feature, or anything that interacts with a phone books.
On Wed, Feb 22, 2012 at 5:02 PM, Josh Patten jpat...@ezuce.com wrote:
I have never heard of anyone accidentally downloading CentOS 6 updates
from doing a simple 'yum update'
I highly recommend updating all your CentOS packages too simply because of
all the security issues that are resolved
On Wed, Feb 22, 2012 at 5:21 PM, Becker, Jesse beck...@sunyulster.eduwrote:
I stopped services.
Updated to 4.4.0- 2012-02-08EST09:10:08 and started services back up.
Still no response using the find command. Other commands such as who, call
(if providing a number) and help work. We also still
On Sat, Feb 11, 2012 at 6:53 PM, Todd Hodgen thod...@frontier.com wrote:
There was a note from one of the Dev’s in the last 60 days or so that said
that 4.6 was actually looking to require less memory, so this should help to
improve things even more.
True. I don't have the results of
Tim, I welcomed your frank email, but I'm weird like that ;)
On Thu, Feb 16, 2012 at 9:02 PM, Tim Ingalls t...@sharedcom.net wrote:
Routing inbound calls to an auto-attendant worked great for a long time and
then just stopped working one day. After connecting the call, visitors were
greeted
testing
and fixes.
Past release changelog
http://download.sipfoundry.org/pub/sipXecs/ChangeLog-4.4.0
Build Log==
commit db499880abdfcba6c0fd957596232eb8b4e666f3Author: Douglas Hubler
dhub...@ezuce.comDate: Wed Feb 8 10:08:36 2012 -0500
build err : from jbaclor patch
On Thu, Feb 16, 2012 at 9:02 PM, Tim Ingalls t...@sharedcom.net wrote:
Hi Everyone. I promise I am not trying to be a troll. I have some serious
questions that I hope I can ask honestly and get some honest feedback about
using the free version of sipXecs as a commercial product.
Tim, it's at
Very valid point.
I've always been of the opinion that things like this you want to use
at both a page level and a section level. Sometime you want to mark
a full page, other times only a specific section.
So on that, note, something like this at the top of a page or at the
beginning of a
On Wed, Feb 15, 2012 at 11:19 AM, Михаил Родионов ma...@siplabs.ru wrote:
Hello,
One of our customers today asked for a feature he *loved* in his old
Panasonic PBX - return of calls being blindly transferred that got no
response after timeout.
Panasonic PBX can act like this:
when someone
On Wed, Feb 15, 2012 at 1:49 PM, Matt White mwh...@thesummit-grp.com wrote:
The current template for voxitas will work with the appia registration
platform.
Voxitas merged with appia about 1-2 years ago, prior to which appia only had
the IP based platform...and voxitas only the registration
On Thu, Feb 9, 2012 at 5:01 AM, pscheep...@epo.org wrote:
I think for a production environment it's much easier to run 4.4 from ISO
(thus Centos 5) and upgrade by
-backup
-ISO install of 4.6
-restore
the theory i guess was that you're changing 2 variables at the same
time (CentOS 5 - 6 and
On Thu, Feb 9, 2012 at 6:21 AM, pscheep...@epo.org wrote:
I thought it was more a question what would be the easiest (as in least
effort) way to move from CUCM to SipX 4.4 and then to 4.6.
I thought Jesse wanted to run 4.4 on CentOS6 just to have an easier upgrade
path.
But if he runs 4.4 on
to run 4.4 on CentOS6 just to have an easier upgrade
path.
But if he runs 4.4 on CentOS6 he's all alone, if he runs 4.4 on CentOS5 he's
part of a big group.
Not sure what his arguments for running 4.4 on 6 are though.
Douglas Hubler dhub...@ezuce.com wrote on 09-02-2012 12:00:44:
On Thu
On Thu, Feb 9, 2012 at 9:36 AM, Becker, Jesse beck...@sunyulster.eduwrote:
It looks like you like staying on Centos 5 and 4.4 is the best transition.
I will plan on installing on the server and then doing an restore from my
backups.
Are you saying then when SipX 4.6 / CentOS 6 comes out that
On Wed, Feb 8, 2012 at 12:04 PM, Becker, Jesse beck...@sunyulster.eduwrote:
My site is making the first step in moving away from Cisco UCM to SipX
ECS. Our IT department has been running several months on SipX 4.4/Centos5
using Polycoms and a Desktop Computer. Our next step is the purchase of
On Wed, Feb 8, 2012 at 12:42 PM, Becker, Jesse beck...@sunyulster.eduwrote:
Thank you for your reply. I was not planning on doing an in place upgrade
from Centos5 to 6. i was wondering if CentOS 6 could be installed on the
new servers and then install SipX 4.4 from RPM and restore the
There were some very fundamental changes in 4.6 that poses the
following question about how admin confirms changes to the system. I
captured situation in this tracker issue:
http://track.sipfoundry.org/browse/XX-10028
It's unclear if we will be able to implement the mockup, it's only
there to
On Sat, Feb 4, 2012 at 8:29 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
In the mockup does it auto fill in the current date/time?
y, i didn't add that to mock up, but real ui would auto fill in just
like phone profiles.
On Sat, Feb 4, 2012 at 8:20 AM, Michael Picher mpic...@ezuce.com
There were a few more changes that didn't get in, but there are the
ones that did. I didn't realize there were so many changes to 4.4 and
according to George, we're waiting on a few more so this won't be
pushed to 4.4 stable quite yet.
commit e19470bb65522aa52845eafbf283ff63f7c3b55b
Author:
On Wed, Jan 18, 2012 at 12:03 PM, Roman Gelfand rgelfa...@gmail.com wrote:
Once logged in, I have noticed that pages use
base href=https://host.mydomain.com:8443/sipxconfig/; /!--[if
IE]/base![endif]--
it is not easily editable. sipxconfig.war has the files and I believe
they are expanded
I'm just now starting to get something working, but still no rpms
thought. Next week, I will have rpms, sorry for the delay
--
Join me to talk about sipXecs and the upcoming version 4.6 at
CoLab @ CSU March (5th 6th).
http://www.sipfoundry.org/sipx-colab
Hack with me on at the CoLab
On Mon, Jan 9, 2012 at 3:31 AM, Todd Hodgen thod...@frontier.com wrote:
Developers – There needs to be a broad discussion about support for Polycom
version 3.3 and beyond, as well as support for the next generation phones
they have or are releasing in 2012. The first phone – the VVX500 is out
On Thu, Jan 5, 2012 at 4:11 AM, Henry Dogger h.dog...@telecats.nl wrote:
Is this normal behavior? A server under this kind of load will eventually
start using its swap memory and then failing (we tested this).
I saw the same behavior. We're talking about RSS or resident memory I
assume.
On Wed, Jan 4, 2012 at 3:05 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
Is 4.6 going to move the CDR database to mongoDB? If so, will there be an
easy way to AUTOMATICALLY export CDR records from mongoDB? I’m curious if
I’ll need to move away from my easy access app via odbc and
On Wed, Dec 7, 2011 at 2:17 PM, Ken Ridley k...@federico.net wrote:
We are not using VOP, 3 users with Polycom 501 phones are part of hunt
groups that the DIDs are directed to
Other systems display the call with the hunt group name
Theses phones were connected to another VIOP system, that
On Tue, Dec 6, 2011 at 4:55 PM, Max Clark max.cl...@gmail.com wrote:
I've been digging through the settings and cannot find anything that
would seem to address this. What should I be looking at?
try turning off external ring in proxy settings (Servers/(server name)/Proxy)
On Mon, Nov 21, 2011 at 4:31 AM, Jeff Ferrara jferr...@rhino.com.au wrote:
Just wondering if anyone knows of an easy way to override the phone book
supplied by sipX to Polycom phones.
I want to change the rt (ringtone) field of the Polycom
-directory.xml file per contact to allow
not sure what happened, but i reset password from user management
(crowd). let me know if this doesn't work
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On Tue, Nov 8, 2011 at 12:48 AM, Kumaran T
thiru.venkateshwa...@ezuce.com wrote:
In user portal-My information-contact
information-Address-office-which contain the Assistant and Assistant
phone field.So these field are present in the Valid tab(Office address)
or otherwise these field should
On Tue, Nov 8, 2011 at 6:22 AM, Venkateshwaran T
thiru.venkateshwa...@ttplservices.com wrote:
I meant in user portal-Address tab(office),Assistant and Assistant
phone field are present.So my doubt is whether these field should be
present in Address Tab only?Because its Office Address tab
On Tue, Nov 8, 2011 at 7:55 AM, Venkateshwaran T
thiru.venkateshwa...@ttplservices.com wrote:
Any update regarding this.
IMO i think it's a trivial issue, sorry. That's my only update.
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List
On Thu, Nov 3, 2011 at 10:32 AM, Jan Fricke jan.fri...@iant.de wrote:
Hi,
I know that there are some end-devices that support CCBS (call completion on
busy subscriber). E.g. we used it some time ago with snom phones. If I’m not
wrong the feature is implemented on phone side with some kind of
In order to effectively release sipxecs 4.6, we need to forgo support
on CentOS 5.
Full story..
Replacing internal database storage FreeDB (10 years defunct FOSS
project) with mongodb has been going fantastic. So far we're seeing
3x performance at proxy and we've reduced the code in many c++
On Tue, Nov 1, 2011 at 9:14 AM, Burleigh, Matt
matt.burle...@eiisolutions.net wrote:
So sipx 4.6 can be a major upgrade since there is no upgrade path from
CentOS 5 to CentOS 6 and will require a complete reinstall of the
operating system. Can a 4.4 backup be restored to a 4.6 without issues?
On Tue, Nov 1, 2011 at 9:52 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Are there any issues with installing centos6 on a machine and installing
sipx 4.4 via rpm methos, so the update is a yum method or at least easier
for planning purposes?
only issue is that sipx 4.4 doesn't build
On Tue, Nov 1, 2011 at 12:19 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
So while I can do a mimimal install of centos 6.0 now, it is probably better
to wait until the freeswitch part is figured out first, correct?
the rpms published to download.sipfoundry.org are built under centos 5
On Tue, Nov 1, 2011 at 3:13 PM, Geoff Van Brunt gvanbr...@dstgroup.com wrote:
I have an problem that may be a known issue. I forgot to do a yum update
after installing the lastest ISO for 4.4.
I had a user and I changed their User ID. Now the old User ID cannot be
reused as it complains that
On Wed, Oct 26, 2011 at 2:22 PM, Bob Anderson b...@callmart.com wrote:
** **
** **
Can anyone please let me know what this means?, system is running fine
Not sure, but maybe you can figure out from here
On Fri, Oct 21, 2011 at 5:15 AM, Jan Fricke jan.fri...@iant.de wrote:
Is OpenACD able to behave like the old acd system configured with answer
mode “Ring all” (one incoming call, all phones with matching skills ring.
Other incoming calls at this time will hear music. After one of the agents
On Fri, Oct 21, 2011 at 8:16 AM, George Niculae geo...@ezuce.com wrote:
Discussing with Douglas and there are two options:
- to remove it altogether, so gw and phones profiles will be never
pushed on upgrades
+1
- control it from /etc/sipxpbx/sipxconfig.properties, a flag like
Change log for past releases can be found here
http://download.sipfoundry.org/pub/sipXecs/ChangeLog-4.4.0
Update #14 : Thu, 20, Oct 2011
==
- ** No security updates in this update **
- ISO has *not* been rebuilt as decided in release
On Wed, Oct 19, 2011 at 5:56 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I see the sipx-dev list is archiving. I do not see any archives after
september 16 of the sipx-users list. is something wrong or is it just my
browser?
that's about the time i had to restart system, it's not
Be sure to update your system to the most recent sipxecs 4.4, there
have been a number of memory leak fixes. One in particular with
sipXimbot that will grow w/o bounds if it cannot connect to registrar
plugin for any reason.
FYI: 4.4's internal database storage (IMDB) was used in anger in a lot
On Fri, Oct 14, 2011 at 8:19 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I am going to suggest that: http://track.sipfoundry.org/browse/XX-5120
Should be rolled back. If the callerid is +(whatever), it's a valid e.164
format and should be dial-able. It should not be removed. It is
continued testing and fixes.
Build Log
==
commit ce96cfb922492ad4d0c5ddf27d9fc1a8fd795ce0
Author: Douglas Hubler dhub...@ezuce.com
Date: Wed Oct 12 05:55:29 2011 -0400
XX-9902 - close file handles that were left open. one place file
handle was not closed because
On Thu, Sep 29, 2011 at 10:05 AM, Niek Vlessert niekvless...@gmail.com wrote:
Is there some tag in Git about this update? So we can do a pull, make a new
jar with some changes based on the right sources, so we can copy it onto a
running binary-only SipX 4.4 update #12 system?
Sorry, i missed
Update #12 : Wed, 5, Oct 2011
==
- ** No security updates in this update **
- ISO has *not* been rebuilt as decided in release policy. Yum update
after installation is recommended for getting these updates.
- Thank you all for your
On Sun, Oct 2, 2011 at 1:38 PM, Becker, Jesse beck...@sunyulster.eduwrote:
All,
I recently purchased two Polycom IP Backlit Expansion Modules for testing.
I have installed on two Polycom IP 650s. However, when configuring speed
dials, it appears that the last button on the BEM never
On Tue, Sep 27, 2011 at 2:18 AM, Alberto g...@interfree.it wrote:
won't be compatible with html config file anymore. I'm working on a new snom
plug-in specifically designed for version 8.x.
Excellent. Be sure to note that phone support moved into separate
projects/rpms in the development
On Tue, Sep 27, 2011 at 6:52 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I think he is suggesting that on newer snom firmware auto provisioning with
sipx will not work unless dhcp.conf is modified. I think he is suggesting an
improvement to the default dhcpd.conf to get around that.
On Tue, Sep 27, 2011 at 12:51 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I wont break anything. We can even add the same code for 0004f2 (polycom) to
be specific to a polycom, otherwise it takes the defaults.
Sounds like patch is a go then
What is unclear
is how to implement it on
please ignore, sorry for noise
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the disk filled up and i'll need to add a bigger drive. i freed enough
space to keep it going for now. expect some outages during the day. if
your post in the last 36 hours didn't go thru, I apologize, you may have to
repost.
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continued testing and fixes.
Build Log
===
commit 0d8b1e2c265cd2c71095034cf020f28fb8a5eabc
Author: Douglas Hubler dhub...@ezuce.com
Date: Thu Sep 8 15:31:18 2011 -0400
XX-9848 - repair infinite loop when uri ends in % with only zero
or one characters
commit
On Fri, Sep 9, 2011 at 9:09 AM, Kumaran T
thiru.venkateshwa...@ezuce.com wrote:
For taking fresh freeswitch.log,I had removed the freeswitch.log
from /var/log/sipxpbx.And then I dialed 100,101,OpenAcd line but the
freeswitch.log was never created automatically.
It will generate
On Fri, Sep 9, 2011 at 12:25 PM, Gary Luca garyluc...@gmail.com wrote:
Just for the hell of it I tried using TCP instead of UDP for SIP
signalling. And it works no problem.
Can you configure the phone manually to do UDP and get inbound calls
to work? If so, can you dump the configuration
I uploaded builds for CentOS 6, but only for development binaries :
0.0.4.5.2. If all tests out ok and I can backport FS fixes to sipxecs
4.4 to support CentOS 6, then sipxecs 4.4 binaries can be built. I
have no idea about CentOS 6 as an ISO at this point.
On Thu, Sep 8, 2011 at 3:38 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I have had no issues with it since installing it as soon as it was
published.
thanks tony.
I'm going to add one more fix to update #10
http://track.sipfoundry.org/browse/XX-9848
I should still get this out
2011/9/4 Tony Graziano tgrazi...@myitdepartment.net:
how polished is the plugin? does it have support for the open on config?
would you consider contributing it to the codebase?
alberto, *bump*
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http://download.sipfoundry.org/pub/sipXecs-stage
- includes following fixes.
===
commit 5b3a195e85c65e3b51f676481180202178588d9f
Author: Mircea Carasel mirc...@ezuce.com
Date: Fri Sep 2 14:30:32 2011 +0300
XX-9836: sipXrest and sipXivr
On Thu, Sep 1, 2011 at 12:29 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Users can have personal speeddials. What I am seeing is there is not
import/export utility for the user.
contact manager for end user that doesn't allow import/export does
seem inflexible. Feature makes sense to
On Wed, Aug 31, 2011 at 3:05 AM, Kumaran T
thiru.venkateshwa...@ezuce.com wrote:
IMO I though that it will take the non-numeric value will be the
preferences but always the user-id will the default IM-id
On the surface, this looks like a neat idea, but it would create all
sorts of
On Tue, Aug 30, 2011 at 9:19 AM, George Niculae geo...@ezuce.com wrote:
Please give it a try in next build and let me know how it works
Thanks Tony and George.
I'll publish a URL to build when it's available.
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and Roman Gelfand and Cyril Constantin
for your bug reports!
Build Log
===
commit 97ba58d4d427dc6a5113a1c18d63caa808f1592e
Author: Douglas Hubler dhub...@ezuce.com
Date: Thu Aug 18 08:34:49 2011 -0400
XX-9811 - add support for custom permissions replication back
On Tue, Aug 16, 2011 at 5:15 PM, Sven Evensen sven.even...@onrelay.comwrote:
** ** ** **
Hi, the itsp is very flexible and expects CLI in from field within the DID
range. So therefore using “transform extension”**
should be straight forward, but it just doesn’t work.
sipXbridge config is
From what i can tell, custom permissions are not getting into
/var/sipxdata/sipdb/permission.xml. I'm looking into it, not obvious
from the commits.
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2011/8/15 Sven Evensen sven.even...@onrelay.com
** ** ** ** ** ** **
This is an issue I have seen now and then and I also saw mentioned earlier
here, but was never resolved.
** **
On outbound calls through SIP trunk to ITSP, we want every user to emit own
CLI. Normally that will
===
commit 94b409b7ca5b51c6eb598f4333b4d31205852f56
Author: Douglas Hubler dhub...@ezuce.com
Date: Tue Aug 2 19:54:30 2011 -0400
XX-9791 - typo causing out of order errors
commit 3a23f51572e0f2fccd19215de0a6468f0c92c84d
Author: George Niculae geo...@ezuce.com
Date: Tue Aug 2 18:47:34
On Fri, Aug 12, 2011 at 8:01 AM, cyril constantin
cyril.constan...@gmail.com wrote:
Call from sip peers on Asterisk to PSTN Avaya works without issue.
Call from Sipxecs sip peers through Asterisk then using H323 connection to
Avaya extension works
Call from Sipxecs sip peers through Asterisk
Unless i missed it, we want to identify the process that's consuming the
memory.
Given the thread dump, i *think* it's openfire. if it is
when it's trying to unify presence. its possible it cannot connect to the
registrar pluggin that feeds it presence change. possible logs on either
side
eZuce team has discovered and fixed a number of bugs in sipxecs 4.4.0 when
testing sipxecs up to 30K users over 5 servers. Previously 10K was
difficult and more than 3 servers was not possible. We'll be merging the
fixes into 4.4.0 code base in next week or so. I wanted to give a heads up
On Mon, Aug 8, 2011 at 12:58 PM, Douglas Hubler dhub...@ezuce.com wrote:
eZuce team has discovered and fixed a number of bugs in sipxecs 4.4.0 when
testing sipxecs up to 30K users over 5 servers. Previously 10K was
difficult and more than 3 servers was not possible. We'll be merging
On Tue, Aug 2, 2011 at 9:56 AM, jnolen jnol...@mindspring.com wrote:
I have the following error on a version 4.2.1-18890.6.1 system:
Alarm: SPX00030
Severity: CRIT
Alarm Text: Process 'CallResolver' failed its configuration test.
psql: FATAL: database is not accepting commands to
On Mon, Aug 1, 2011 at 5:28 PM, Aaron Pursell aar...@esgw.org wrote:
How? By creating a new set of device files using the EXACT SAME files and
deactivating the current ones and activating the new ones. I seriously
downloaded the files from the SipX GUI and then created a new device files
set
On Fri, Jul 29, 2011 at 10:54 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
#4 and parts of #5 really point to no qos or shaping on the internet
connection. if ntp cant sync in a reasonable period, clocks wont sync.
sounds like an isp/shaping issue.
re:ntp
If time on devices starts to
On Fri, Jul 29, 2011 at 11:00 AM, Aaron Pursell aar...@esgw.org wrote:
I did test all that dns is working. It finally started
working as you replied I have to restart the services 2-3
times to finally get it to work.
I've been there too. There's a caching DNS server on machines that
often
On Mon, Jul 18, 2011 at 12:44 PM, Henry Dogger h.dog...@telecats.nl wrote:
When my status is set to DoNotDisturb (I have checked the setting, send to
voicemail on DND) I can’t use the function pickup or listen on the mybuddy.
In fact every function concerning the MyBuddy calling my phone is not
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