On Mon, Feb 14, 2011 at 4:47 PM, Geoff Van Brunt gvanbr...@dstgroup.com wrote:
Jira does work OK as a Roadmap but it requires a lot of reading to find
out the details. I can use Jira, but not everyone can... You have to
read through an awful lot of entries to find out what is included in new
On Fri, Feb 11, 2011 at 5:33 PM, Jim Canfield jcanfi...@emstar.com wrote:
Finally have a working Patton 5200 SBC config working on Voip.ms as B2BUA.
Here's a rough draft template for anyone who might be interested.
There's a great place for your notes in a page added off of this page.
On Thu, Feb 10, 2011 at 8:01 AM, Ben Goodfellow b...@btg-computers.co.ukwrote:
I just tried to update 4.4.0 using the yum update command and receive the
following error – can anyone shed any light?
-- Finished Dependency Resolution
dirac-1.0.2-1.el5.rf.x86_64 from installed has depsolving
On Thu, Feb 10, 2011 at 9:27 AM, Gmb sipxm...@gmail.com wrote:
Hi all,
has anyone tried Audiocodes 3x0HD IP phones?
Is the provisioning mask of these phone in roadmap? I haven't found
nothing about that...
i have a 320HD i'd love to put to use one day. only thing i've heard
negative was that
6 guys:joegen, laurentui, cristi, george, mircea and myself are going
to spend two full days updating the wiki. We'll be coordination in
IRC #sipx on freenode if you're interested.
Also, I'll be upgrading confluence tonight in hopes it fixes the
indexing problem.
On Tue, Feb 8, 2011 at 2:25 PM, Bob Blanchard Jr. bl...@dainty.ca wrote:
Regarding setting up a truly bilingual voicemail system, we found this post:
http://www.mail-archive.com/sipx-dev@list.sipfoundry.org/msg01992.html
But we were not able to get this to work at all.
What are the details
On Tue, Feb 8, 2011 at 6:08 PM, McIlvin, Don
don.mcil...@nrtnortheast.com wrote:
FYI – Just thought I would mention the following solution for those that
have Polycom Productivity Suite and want to enable reports (i.e. turn it
on).
The RTCP-XR-Enable.cfg file simply sets
On Mon, Feb 7, 2011 at 3:40 PM, Michael Scheidell
michael.scheid...@secnap.com wrote:
as an administrator, I would like to know lusers who let their voicemail box
get full.
define full
How do I do that?
depends on answer above
logs? alerts.
Ultimately it seems alerts would provide the
On Fri, Feb 4, 2011 at 4:49 AM, Peter van der Salm
peter.vanders...@smart-future.nl wrote:
Then SipXproxy sends and INVITE to sipXbridge to setup the call to
0642769062
sipXbridge returns a 500 Server Internal Error.
I would think /var/log/sipxpbx/sipxbridge.log should have an error message.
,
3572 ZR Utrecht,
The Netherlands
phone: +31 308 793 512
fax: +31 847 156 296
mobile: +31 620 749 471
petervanders...@smart-future.nl
On Feb 4, 2011, at 14:13 , Douglas Hubler wrote:
On Fri, Feb 4, 2011 at 4:49 AM, Peter van der Salm
peter.vanders...@smart-future.nl wrote
On Thu, Feb 3, 2011 at 3:02 PM, Burleigh, Matt
matt.burle...@eiisolutions.net wrote:
I’ve been experimenting with 4.4 and it seems that all of my Polycom phones
are acting differently then they did with 4.2.1. When a calls comes in I
used to be able simply pickup the handset and begin speaking
On Mon, Jan 31, 2011 at 3:01 PM, Burleigh, Matt
matt.burle...@eiisolutions.net wrote:
Is it possible to allow a user permission to add a phone to their account?
We had this functionality in ShoreTel where a user can assign his extension
to a hard phone. And do this from a phone. Is this
, but I wrote guides some time back. I'll work on restoring that page
eventually...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Monday, January 31, 2011 3:26 PM
To: Discussion list
Wrong list and project.if you ultimately find the right list let me know so
I can help other lost folks
On Jan 31, 2011 8:26 PM, Li, Yvonne yvonne...@teamaol.com wrote:
sipxCallInjectMediaPacket I guess.
From: Li, Yvonne
Sent: Monday, January 31, 2011 3:15 PM
On Fri, Jan 28, 2011 at 10:01 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
I seem to be one of the heavier ACD users. I would definitely like to
test this out in the latest version and see if I can provide helpful
feedback.
Is the ACD role the old ACD and Call Center
On Sat, Jan 29, 2011 at 8:00 PM, m...@grounded.net m...@grounded.net wrote:
I generated a CSR file using the GUI but it's only 1024bit while godaddy only
accepts 2048.
really? hmm, we should bump this up then, please file a bug
I re-created my file using the correct bit size and got my
On Thu, Jan 27, 2011 at 10:44 AM, Worley, Dale R (Dale)
dwor...@avaya.com wrote:
In regard to the message outgoing call 1, I believe that is a debugging
message that was incorrectly set to log at ERR level rather than DEBUG
level. A programmer could probably look at the code and determine if
On Thu, Jan 27, 2011 at 10:16 PM, srinivasa rao ssv...@yahoo.com wrote:
2) Afterthis, I called bandwidth again; bandwidth asked me to change a
value of param name=inbound-codec-negotiation value=greedy/ in the
freeswitch/conf/sip_profiles/internal.xml file. Therefore, I rebooted the
server.
On Fri, Jan 28, 2011 at 4:24 AM, Nikolay Kondratyev k...@nstel.ru wrote:
Hi, all,
I continued to investigate my problem.
And i found that when a call with that bloody ;phone-context=cdp.udp in
the user part of request uri is coming to hostname (which is configured as
sip domain alias), that
On Fri, Jan 28, 2011 at 5:58 PM, Charles Chalekson chalek...@gmail.com wrote:
I had not noticed even number version labeling as beta previously. I always
thought developmental unstable versions were odd numbered and when once
stable was changed to an even number. I guess I know now.
On Fri, Jan 28, 2011 at 7:25 PM, srinivasa rao ssv...@yahoo.com wrote:
could you please tell me the file names for the template files?
every file in that directory (not sub directories) is a template.
they end in .vm
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On Fri, Jan 28, 2011 at 8:25 PM, Wayne A. Green
wgr...@bbtconsulting.com wrote:
I have a very strange problem associated with the auto-attendant responding.
I am currently running (4.2.1-018971.dhubler 2010-08-21T04:59:18 build34).
This is a new installation and has been operational for
On Fri, Jan 28, 2011 at 5:03 PM, Charles Chalekson chalek...@gmail.com wrote:
Attempting to yum update to 4.4.0 from 4.2.1 and am getting a failure due to
some dependency issues.
Am I doing something wrong?
Thanks,
Charles
-- Running transaction check
--- Package erlang.i386
On Fri, Jan 28, 2011 at 5:52 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
No it actually doesn't always.
That is why it is a snapshot development version.
I've had to start over several times myself. That is expected in a
non-production release cycle.
Let's not muddy the waters
great question. short answer is it must be INVITE, not REFER as it
stays in the media path.
Looking for a REFER based connection to agent would allow for some
insane scalability as OpenACD really only consumes a fraction of the
CPU to do it's job. OpenACD today uses the FS session to determine
On Mon, Jan 24, 2011 at 5:51 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
it sounds like an internal dns issue. I think there is a JIRA on this.
How long before the local phones stopped being able to register, 30 minutes?
I found the DNS advisor found in the System menu fairly useful
On Mon, Jan 24, 2011 at 10:17 AM, m...@grounded.net m...@grounded.net wrote:
Looks like a git file, looks like I run something like 'git apply patchname'
but don't want to break my system so can't run it til I know more.
I did manage to find a little information but it's like taking on a
So glad your checking out ACD, thanks! comments in-line
On Mon, Jan 24, 2011 at 1:16 PM, andrewpit...@comcast.net wrote:
I'm running the 4.4.0-53 build from the SIP Foundry site, and it seems that
the autoattendant gets broken for remote workers when adding dial plans to
support OpenACD
On Fri, Jan 21, 2011 at 6:32 AM, Rama Krishnam Raju Pakalapati wrote:
Thanks for the information, i browsed the wiki pages you have referred and
it points to the
http://article.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/29381
link and explains the cases tested.
I have some more
As a workaround, If you disable ACD as a role in the server
configuration menu option, then that process shouldn't start.
On Thu, Jan 20, 2011 at 8:09 AM, Nikolay Kondratyev k...@nstel.ru wrote:
Hi, all.
I started to play with 4.4.0.
Today I upgraded my test system from 4.2.1 to 4.4.0 (4.4.0-
On Thu, Jan 20, 2011 at 5:28 AM, Henry Dogger h.dog...@telecats.nl wrote:
Hm, I tried making a dummy user for the mediant, which has permissions
to call everything.
The mediant registered using this dummy user, I can see it is registered
in the registrations list.
But somehow my calls keep
On Wed, Jan 19, 2011 at 9:14 AM, Rama Krishnam Raju Pakalapati
ramud...@gmail.com wrote:
I hope the implementation of account code xx-4824 mentions that, the
authorization code feature was designed for Outgoing calls and the use case
is similar to the case what we are testing..
It should also
Technically upgrades from dev releases are not supported but if you get past
yum update and system runs you should be good going forward
On Jan 19, 2011 6:37 PM, Rolland Hart rolland.h...@gmail.com wrote:
Can this version of Sipx (sipXconfig (0.4.4-5b9dc0c 2010-12-01T21:31:52
build32)) be
On Tue, Jan 18, 2011 at 5:49 AM, Michal Bielicki
michal.bieli...@seventhsignal.de wrote:
If you wait another week I'll throw in a fresh mod_smp in freeswitch ;)
Good news is freeswitch is independent now (no signifigant
modification made for sipxecs) so folks should be able to update
freeswitch
On Tue, Jan 18, 2011 at 7:54 PM, Richard Bruce
rnbr...@dimensionalcom.com wrote:
Can someone direct me to where the dial by name feature is
controlled?
https://github.com/dhubler/sipxecs/blob/master-4.2/sipXivr/src/main/java/org/sipfoundry/sipxivr/DialByName.java
and accompanying files.
I want to do a full build sometime tomorrow and run a smoke test. Then
we can declare beta unless anyone has any concerns on what they've
seen so far.
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Why can't you just ignore the user group?
On Jan 16, 2011 9:49 AM, m...@grounded.net m...@grounded.net wrote:
Are there any other places to look for information about the LDAP
functions? I've spent countless hours searching and can't find the answers
to what I'm trying to achieve. I've been using
On Sat, Jan 15, 2011 at 8:50 PM, Richard Bruce
rnbr...@dimensionalcom.com wrote:
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To: AANLkTimsXJwRZVowEAGo=7Hi6c+6+MbT8PYJ1Xqjo=x...@mail.gmail.com
X-FUDforum:
On Fri, Jan 14, 2011 at 4:47 AM, Nikolay Kondratyev k...@nstel.ru wrote:
Hi all,
i found that by default sipxconfig plugin sets Cfwd Busy Dest incorrectly.
By default the value is set to 'vm'. This leads to incorrect behaviour when
DND mode is activated on spa942 phone.
The thing is that,
also check the freeswitch log,
On Wed, Jan 12, 2011 at 9:20 AM, Huw Jones hgw.jo...@llandrillo.ac.uk wrote:
Thank you (and Tony) for the suggestions. I'll investigate as per your advice.
Cheers
Huw
Nikolay Kondratyev 01/12/11 11:08 AM
Somebody (something :) ) from your local lan
On Mon, Jan 10, 2011 at 6:47 PM, Bob bobsjunkm...@bellsouth.net wrote:
which date is wrong for you and what should it be? what is your timezone?
It's not the date, but the time of the call as played in the TUI. The time
displayed on the web interface is correct for the messages.
If I recall
On Mon, Jan 10, 2011 at 10:48 AM, Geoff Van Brunt
gvanbr...@dstgroup.com wrote:
I agree. An install all and disable unneeded is bad from a security
perspective as well. As we all know SIP products are in the crosshairs
of hackers everywhere with recent automated attacks. This is only going
to
On Sat, Jan 8, 2011 at 2:15 PM, Bob bweybre...@bellsouth.net wrote:
I'm running 4.2.1 under RHEL. Basically installed it via YUM, basic
configuration options, etc. The problem is this:
If you listen to a voicemail received at 10:00 AM Eastern time, if you play
the message info in the TUI, it
I will upload them tonight. I noticed build didn't include them and I meant
to upload them
On Jan 8, 2011 7:37 AM, Matt White mwh...@thesummit-grp.com wrote:
Douglas Hubler dhub...@ezuce.com 01/07/11 11:58 PM
There are still outstanding bugs
http://bit.ly/gCCsCX
CentOS repo file
There are still outstanding bugs
http://bit.ly/gCCsCX
CentOS repo file
http://download.sipfoundry.org/pub/sipXecs/sipx-4.4.0-centos.repo
ISO (look for file labeled 4.4.0)
http://download.sipfoundry.org/pub/sipXecs/ISO/
(ignore openacd ones, they are slated for next release) but we've done
On Thu, Jan 6, 2011 at 4:39 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
I've not had the pleasure of using tomcat/java servlet at all - but I'm
assuming that if one knew where to find the servlets for the page, you could
modify the code to check to see if that user is part of a
On Thu, Jan 6, 2011 at 7:34 PM, m...@grounded.net m...@grounded.net wrote:
Bloating? I don't consider sipx bloated, but everyone has an opinion (me
too).
Not saying it is bloated, saying not to let it go that way.
it's all about perspective. I think sipxecs is on the heavy side, but
i
On Fri, Jan 7, 2011 at 12:34 AM, Roy Walker roy.walker...@gmail.com wrote:
Hate to dig up an old thread, but it looks like I ran into
this as well... strange thing was that the digit map was in
the config, but the phone did not use it (Polycom 650
running 3.2.3). I manually put the digit map
On Wed, Jan 5, 2011 at 9:15 AM, Henry Dogger h.dog...@telecats.nl wrote:
A client of ours is having problems with the sipXecs config ui, it is very
instable and gives a lot of internal server errors.
I would review the /var/log/sipxpbx/sipxconfig.log for meaningful
error messages. If you
effectively gone. http://sipx-wiki.calivia.com. We can put in
global redirect to new wiki wiki.sipfoundry.org.
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On Wed, Jan 5, 2011 at 4:40 PM, srinivasa rao ssv...@yahoo.com wrote:
We have auto attendants setup in our sipx setup. For the last few weeks, these
auto attendant messages are played very very fast. For example, our main
message
used to take 40 seconds to complete it, and it takes very less
On Tue, Jan 4, 2011 at 8:43 AM, Matt White mwh...@thesummit-grp.com wrote:
Douglas Hubler dhub...@ezuce.com 12/30/10 1:55 PM
Building using mock is currently on 4.5 codebase so instead of
backporting it, I recommend we cut the new 4.4 release from 4.5 HEAD
and begin to stabilize this branch.
I
On Tue, Jan 4, 2011 at 11:07 AM, Douglas Hubler dhub...@ezuce.com wrote:
On Tue, Jan 4, 2011 at 8:43 AM, Matt White mwh...@thesummit-grp.com wrote:
Douglas Hubler dhub...@ezuce.com 12/30/10 1:55 PM
Building using mock is currently on 4.5 codebase so instead of
backporting it, I recommend we cut
There is a bit of a delay on 4.4 for at least 2 of the following reasons:
1.) Switch from using OBS to mock for building rpms
2.) FreeSWITCH in a bridge setup address call transfer to AA/VM but
opened a slew of other issues ezuce it still working on
ezuce developers were working on these two
On Tue, Dec 28, 2010 at 9:15 AM, Eda Ercan edaer...@gmail.com wrote:
The owner of sipx installation directory is a user other than sipxchange, it
is the user that makes the build process, but everyone has the rights to
read/write (777) - does it matter?
Couple things
1.) CDR report processing
On Thu, Dec 23, 2010 at 12:03 PM, Alexander Shvaryov shvar...@gmail.com wrote:
Apologies for the typo, I meant ... other than the 711U?
I'm not 100% positive, but i think the INVITE is generated here.
fix is in source control, but there's no build yet. i will let folks
know when it's ready. Yes, automating build is still WIP.
On Thu, Dec 16, 2010 at 11:22 AM, Cristi Starasciuc
cstarasc...@ezuce.com wrote:
Hi,
The fix for fax is in, and ready to be tested.
Regards,
C
On 11/26/2010
On Thu, Dec 16, 2010 at 4:46 AM, Xavier D. magicrhe...@ouranos.be wrote:
Hello guys,
Does someone already make a 'passive' installation ?
I mean without any interactions with the machine, only by scripting all the
installation (adding servers, configuring them, ...) ?
I'm interested in any
i'm looking into it.
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ok, appears to be ok now. let me know if you have any problems, out of
the ordinary that is
On Mon, Dec 13, 2010 at 9:49 AM, Douglas Hubler dhub...@ezuce.com wrote:
i'm looking into it.
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On Fri, Dec 10, 2010 at 11:28 AM, Henry Dogger h.dog...@telecats.nl wrote:
I also tried to see what’s wrong with sipxproc. Are services are disabled,
when I try to start one it’s get the state: “ConfigurationMismatch”
When I get the statusmessage is contains: [version.mismatch: software
On Wed, Dec 8, 2010 at 9:45 AM, Henry Dogger h.dog...@telecats.nl wrote:
Unlike stated in issue: http://track.sipfoundry.org/browse/XX-8476 that a
issue around these errors is fixed.
I still see these lines in the logging: ERROR: duplicate key violates unique
constraint cdrs_call_id_unique
On Wed, Dec 8, 2010 at 4:30 PM, Joe Micciche jmicc...@redhat.com wrote:
My blackberry doesn't do that (trim). I can delete all or nothing.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
So the other x-hundred or thousand of us have to just deal
On Tue, Dec 7, 2010 at 3:48 AM, Henry Dogger h.dog...@telecats.nl wrote:
Thanks this works :)
But I wonder how the search index needs a rebuild.
Is it something I did, or is it a common problem?
yes, it was unfortunately common and fixed in upcoming release
On Mon, Dec 6, 2010 at 7:59 AM, Arman Güngör armangun...@gmail.com wrote:
I have some phones which support only SIMPLE presence protocol. How can they
exchange presence information with sipX. Is there something that converts
SIMPLE to XMPP? Or is there any other way to use them with sipX?
I
On Mon, Dec 6, 2010 at 2:00 PM, Levend Sayar levendsa...@gmail.com wrote:
Hmm,
That is not the case for me. When I skipped error screen, it just installed
CentOS itself, not the sipxecs on it.
I had to install sipxecs manually, in very cumbersome way.
in isolinux.cfg that's put on the CD,
On Thu, Dec 2, 2010 at 3:32 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
We had an important conference call hosted by sipx this morning –
auto-record had been working in the past – today however – the conference
call did not record. I’ve tried changing users/resetting the auto
On Wed, Dec 1, 2010 at 8:55 PM, Kris Amy kris@nextdc.com wrote:
When will this patch be integrated? Is it possible to patch the current
4.2.1 .jar (so I can rollback to stable as opposed to 4.3.2).
4.2.1 is latest stable, so yes, i can be applied to 4.2.1. may take
me a few days to get to
On Fri, Nov 26, 2010 at 12:52 PM, Burden, Mike m...@lynk.com wrote:
[yum.repos.d]# yum update sipxecs
Loading fastestmirror plugin
Loading downloadonly plugin
Loading basearchonly plugin
Loading mirror speeds from cached hostfile
* sipXecs: download.sipfoundry.org
On Tue, Nov 23, 2010 at 2:31 AM, Raymond phuay...@yahoo.com wrote:
I am trying your method to create SBC for my two eyebeam
to uee TLS but still failed. As i am not expert at creating
SBC, is it possible that you could guide me?
For your info, i had a SIPX server of IP address
10.66.15.8.
On Tue, Nov 23, 2010 at 10:28 AM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
I'm running I'm running sipXconfig (4.2.1-018971 2010-08-17T02:20:18
build20) 64 bit ISO build.
I rebooted last night to resolve a java/CPU issue.
There have been several threads started related
On Tue, Nov 23, 2010 at 10:57 AM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
On 11/23/2010 9:38 AM, Douglas Hubler wrote:
On Tue, Nov 23, 2010 at 10:28 AM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
I'm running sipXconfig (4.2.1-018971 2010-08-17T02
On Sat, Nov 20, 2010 at 9:34 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I think there is a need to fix the indexing of the current site and the wiki
cleanup day was a first strp in cleaning up the wiki.
yes, if you think it's bad now, you should have seen what it was.
The wiki. Web
On Sun, Nov 21, 2010 at 8:11 PM, jun.wen.sipx jun.wen.s...@gmail.com wrote:
Hi, I made a fresh clone in my lab system by git clone
git://github.com/dhubler/sipxecs.git sipx. While the compiling was stopped
by errors of in sipXtackLib as attachment. These errors were never detected
in my
On Fri, Nov 19, 2010 at 1:03 PM, Jeff Gilmore j...@thegilmores.net wrote:
Our ISP recently had a problem where their uplinks to the wider internet were
down. All of the networks at my local site were fine. During this incident,
I tried to use the Sipx GUI on my local server and was unable
On Thu, Nov 18, 2010 at 7:32 AM, Jim Canfield jcanfi...@emstar.com wrote:
I'm not sure how much work is actively being done (tracker seems
vacant), but I'd like to give things a push in the right direction.
How many community members (and Ezuce) are willing to collaborate on a
focused effort
On Mon, Nov 15, 2010 at 7:11 AM, Jesse Reynolds je...@va.com.au wrote:
This thread was very useful to me when upgrading a ISO installed 4.0.4
system to 4.2.1... I had to make a couple of changes to get it to work
though:
On 13/09/2010, at 6:06 PM, Michael Picher wrote:
The steps I use are:
Many folks have complained they get locked out and it's not obvious
what the issue is. If you give me your username i can unlock your
account. I haven't looked into why this is happening, for some reason
it thinks you've made too many failed login attempts, yet this has
happened to people that
On Fri, Nov 12, 2010 at 6:06 PM, Matt White mwh...@thesummit-grp.com wrote:
Just an FYI to everyone. Our ISP serving the webserver that host the forums
pulled a switcheroo on us and changed our ip block without warning.
Anyways, DNS should be updated shortly to reflect the new ip.
changed
On Thu, Nov 11, 2010 at 11:17 AM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
*polite bump request* (if there is such a thing)
We are still having users receiving blank voice mails when the messages are
forwarded from other mailboxes.
I still don't know what build 34
On Thu, Nov 11, 2010 at 3:51 PM, Michael Scheidell
michael.scheid...@secnap.com wrote:
On 11/11/10 3:47 PM, Tran, Ly wrote:
If you are using Cisco 7960s remotely without this issue, can you please
post a sample of your config for these phones. Thanks!
I worked for a LONG time trying to
On Wed, Nov 10, 2010 at 10:16 AM, George Niculae geo...@ezuce.com wrote:
Hi All,
I am working on OpenACD integration (it use Freeswitch dial plans) and
one requirement is to replicate Freeswitch dial plan on admin action
only - say admin configures several extensions and wants this to take
-boun...@list.sipfoundry.org] On Behalf Of Douglas
Hubler
Sent: Wednesday, November 10, 2010 9:24 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] when to reload Freeswitch dial plans
On Wed, Nov 10, 2010 at 10:16 AM, George Niculae geo...@ezuce.com
wrote:
Hi All
On Wed, Nov 10, 2010 at 11:10 AM, Sven Evensen sven.even...@onrelay.com wrote:
We have a problem with the snom, for the version 320, we need an http server
to make available other languages.
Is it possible to use the SipXecs server as an http server ? If yes, where
is the root located ?
On Wed, Nov 10, 2010 at 10:44 PM, Jeremy Fluhmann fluhm...@gmail.com wrote:
Looks like 0.71 has the Freeswitch integration added. But if my guess is
right, would it need to use a Freeswitch install on the local server rather
than potentially connecting with the one on sipXecs? I'll be curious
Everyone,
We had a successful wiki cleanup day although there is still much more
to do. Thanks to all the participated. Here are the details
There was more duplication information that even i expected. This was
mostly because pages could not be found and so new pages were created.
This issue
I'll get in the #sipx IRC channel on freenode.net around 3AM EDT and
kick off the event with details. Don't worry if you miss it, I'll
help people get started no matter when they join.
NOTE: 3AM EDT is GMT +4 or 7 AM GMT
Things you can do to prepare:
1.) If you plan on making a fairly
On Thu, Nov 4, 2010 at 11:47 AM, Dan White dwh...@citadelitg.com wrote:
In version 42 of sipx, can you change the web front end to
your colors, logo, and put links on the page?
you were able to do this before, but I can not seem to find
any of the pages that I need to change.
On Tue, Nov 2, 2010 at 6:39 AM, Abdul Mayat abdul_ma...@hotmail.com wrote:
I think the answer is no, but i though I would double
check!
Is it possible to localize the IVR (specifically voicemail)
by group (or any other parameter) so the voice prompts
played out are specific to the users
Its having a problem. Looking into it.
On Nov 3, 2010 8:21 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
--
==
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.326.5325
Email: tgrazi...@myitdepartment.net
On Wed, Nov 3, 2010 at 10:03 AM, Eelco Brölman e.brol...@telecats.nl wrote:
Hi all,
In our HA setup, all the slave/secondary nodes have some file system
permissions which seems odd to me:
drwxr-xr-x sipxchange sipxchange /
drwxr-xr-x sipxchange sipxchange /etc
drwxr-xr-x sipxchange
On Wed, Nov 3, 2010 at 7:12 AM, Tony Graziano
tgrazi...@myitdepartment.netwrote:
You cannot use a domain alias with xmpp.
Ditto's on the Counterpath statement to me, they have no intention to
support SRV for XMPP, nor would they take it under advisement.
BEGIN HIJACK
Speaking
On Wed, Nov 3, 2010 at 5:27 PM, Sven Evensen sven.even...@onrelay.com wrote:
Where can I download the language packs for sipX 4.2. All I find are some
broken links
like http://sipxecs.sipfoundry.org/rep/sipXecs-il8n/
I have this dir
http://download.sipfoundry.org/pub/sipXecs/Localization/
On Wed, Nov 3, 2010 at 11:13 PM, Joegen Baclor jbac...@ezuce.com wrote:
This is a one liner fix in the relay code.
for (int i = config.getPortRangeLowerBound(); i config
.getPortRangeUpperBound(); i++) {
try {
DatagramSocket sock = new
On Tue, Nov 2, 2010 at 6:29 AM, Abdul Mayat abdul_ma...@hotmail.com wrote:
For the clean up day on Friday, what level of
experience/knowledge of SipX is required? I would like to
volunteer but do not have any knowledge of coding and
wouldnt say I was an expert, but I have deployed a few
On Tue, Nov 2, 2010 at 10:35 AM, Gabe Casey gca...@franklinamerican.com wrote:
Is there any setting changes that would need to be put in so that the portal
understands and accepts the requests from the proxy. We had this working and
after a reinstall could not identify the issue. Links to dojo
In preparation for wiki clean-up day on friday, I'm combining all the
pages into a simple space. The way it was organized into multiple
spaces was not conducive to maintenance. Unfortunately this changes
the URL for everything. There be a lot more changes, i just wanted to
give everyone a heads
i'm not an expert, but i think you have to setup a proxy and not a rewrite.
On Mon, Nov 1, 2010 at 3:27 PM, Gabe Casey gca...@franklinamerican.com wrote:
is there a setting or ssl crt that needs to be adjusted when doing an
external rewrite of the url to the end user portal from outside the
On Sun, Oct 31, 2010 at 2:39 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
The are many changes upcoming in the LDAP import function, but phonebook is
not among them as I recall.
That is correct. Many phones like polycom support ldap directly for
phonebooks. Which phone are you
Developers put together a significant build yesterday, rpms are
updated. Here is the repo file
http://download.sipfoundry.org/pub/sipXecs/sipxecs-centos-snapshot.repo
There is currently a build problem with freeswitch on fedora, we'll
get that worked out eventually. Anyone wants to take a
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