I pushed a new build of the development snapshot and i had to change
repo file a but to reflect the version in the 4.3 series better.
http://download.sipfoundry.org/pub/sipXecs/sipxecs-centos-4.3-testing.repo
Summary of changes (commits made on 2010-10-08 and before)
http://github.com/dhuble
Tony I followed your instructions and it let me in. I dumped the data base
and recreated it and it cleared the problem. Thanks for you time and help.
On Fri, Oct 8, 2010 at 8:54 PM, Rolland Hart wrote:
> I found it . Thanks Tony! I think this is what you need. If you need more
> of the log I wil
I found it . Thanks Tony! I think this is what you need. If you need more of
the log I will send.
On Fri, Oct 8, 2010 at 8:46 PM, Tony Graziano
wrote:
> I mand
>
> /var/log/sipxpbx/sipxconfig.log
>
>
> On Fri, Oct 8, 2010 at 8:33 PM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> it
The last time I saw this I thought it had been corrected. To get around this
the last time I saw it I had to dump the db and start over:
# stop sipxecs if it's running
sipxconfig.sh --database drop create
# start sipxecs and resetup your system.
but it would be good to see the log
On Fri
I mand
/var/log/sipxpbx/sipxconfig.log
On Fri, Oct 8, 2010 at 8:33 PM, Tony Graziano
wrote:
> it will cut a log entry to:
>
> /var/log/sipxconfig.log
>
> grab the offending entry and post it here.
>
>
> On Fri, Oct 8, 2010 at 8:21 PM, Rolland Hart wrote:
>
>> Hey Tony,
>> Rebooted 3 times and tr
it will cut a log entry to:
/var/log/sipxconfig.log
grab the offending entry and post it here.
On Fri, Oct 8, 2010 at 8:21 PM, Rolland Hart wrote:
> Hey Tony,
> Rebooted 3 times and tried three different browsers withe the same result.
> Anything else I can try?
>
> On Fri, Oct 8, 2010 at 8:04
Hey Tony,
Rebooted 3 times and tried three different browsers withe the same result.
Anything else I can try?
On Fri, Oct 8, 2010 at 8:04 PM, Tony Graziano
wrote:
> Reboot
>
> On 10/8/10, Rolland Hart wrote:
> > Hi all,
> >
> > I was wondering if any one else got this error message after updatin
Reboot
On 10/8/10, Rolland Hart wrote:
> Hi all,
>
> I was wondering if any one else got this error message after updating to
> 4.3.2 from 4.3.0? Was also wondering if there is a fix? This is the error
> message:
>
> HTTP ERROR: 500
>
> could not execute query; nested exception is
> org.hibernate
Hi all,
I was wondering if any one else got this error message after updating to
4.3.2 from 4.3.0? Was also wondering if there is a fix? This is the error
message:
HTTP ERROR: 500
could not execute query; nested exception is
org.hibernate.exception.SQLGrammarException: could not execute query
R
Oops...
Looks like my video from CamStudio is a bit messed up...should have tested that
before using it...not to mention I had it shooting a frame every 5 ms at 1920 x
1200...I tried saving it to my file server and it copied a 4 gig file then
crapped out...There is still some video - but it is
Sorry if you have done this, but do you have any log entries that occur when it
fails? Can you see what logs are modified when it happens and see what the
entries are?
--Original Message--
From: Ryan
Sender: sipx-users-boun...@list.sipfoundry.org
To: sipx-users@list.sipfoundry.org
ReplyTo
I think you will have to explain the issue in more detail. I for one have
no idea what you are reporting here. Is this a problem where you can't
change the voicemail active greeting via their telephone extensions? IS
there an error tone, is there a hangup?
Please explain the steps you are takin
I'd have to test it. I would suggest sending the server profiles first if
possible.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 4
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <52993>
Message-ID:
I upgraded my 4.2.0 box to 4.2.1 and the problem is still
there. However, I have a 2nd box that was running 4.
From: Staffan Kerker [ietf-li...@kerker.se]
Note, this is a slightly different call scenario due to some changes in my
setup, but the problem is the same. The call is initiated from an incoming SIP
trunk, forks to two phones behind NAT, one answers (200 O
On 10/8/10 5:03 PM, Todd Hodgen wrote:
Per voip.ms
*[Peter]* If the did is in per minute, we can assure it will come with
at least 10 incoming channels. If your traffic requires it we can
increase, also some dids come with more thatn 10 by default
*[Peter]* If your DID is in per minute plan
Nathaniel, run my shaping script!~
On Fri, Oct 8, 2010 at 5:03 PM, Todd Hodgen wrote:
> Per voip.ms
>
> *[Peter]* If the did is in per minute, we can assure it will come with at
> least 10 incoming channels. If your traffic requires it we can increase,
> also some dids come with more thatn 10
Per voip.ms
[Peter] If the did is in per minute, we can assure it will come with at
least 10 incoming channels. If your traffic requires it we can increase,
also some dids come with more thatn 10 by default
[Peter] If your DID is in per minute plan do not have to worry, as we can
increase channels
On 10/8/10 10:29 AM, Nathaniel Watkins wrote:
All,
Our 911 director has asked me to perform a capacity test for conference calling
utilizing sipXecs. I am trying to get at least 50 participants to join a
conference call so I can monitor bandwidth/load/etc. to see if it is feasible
for us to
I've got some screen capture software running - if there is a reasonable turn
out - I'll record the audio/video and upload it...
Hopefully I'll see some people on there...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Worley, Dale R
(Dale)
Sent: Friday, October 08, 2010 11:20 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Is there a response code tha
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rene Pankratz
[rene.pankratz.l...@iant.de]
Thanks a lot for your answer... though it is somehow nonsatisfying...
Y
Thanks a lot for your answer... though it is somehow nonsatisfying...
René
2010/10/7 Worley, Dale R (Dale)
>
> From: sipx-users-boun...@list.sipfoundry.org [
> sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rene Pankratz [
> rene.pankratz.l...@iant
please record it and put it on youtube ;)
SCNR
Am 08.10.2010 16:40, schrieb Philippe Laurent:
> No problem. I'll call at that time.
>
> On Fri, Oct 8, 2010 at 10:29 AM, Nathaniel Watkins
> mailto:nwatk...@garrettcounty.org>> wrote:
>
> All,
>
> Our 911 director has asked me to perform
No problem. I'll call at that time.
On Fri, Oct 8, 2010 at 10:29 AM, Nathaniel Watkins <
nwatk...@garrettcounty.org> wrote:
> All,
>
> Our 911 director has asked me to perform a capacity test for conference
> calling utilizing sipXecs. I am trying to get at least 50 participants to
> join a conf
On 10/8/10 10:29 AM, Nathaniel Watkins wrote:
least 50 participants
Hope you got 50 CCP's.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
> *| *SECNAP Network Security Corporation
* Certified SNORT Integrator
* 2008-9 Hot Company Award Winner, World Executive
All,
Our 911 director has asked me to perform a capacity test for conference calling
utilizing sipXecs. I am trying to get at least 50 participants to join a
conference call so I can monitor bandwidth/load/etc. to see if it is feasible
for us to host this type of conference call internally.
T
2010/10/8 Laurentiu Ceausescu :
> Should I make a task to update all the users without SIP password?
i would not. Easy workaround is to delete users are reimport.
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfo
Regarding http://track.sipfoundry.org/browse/XX-8936 ...
If the user is new and sip password is null ("SIP password" attribute is not
mapped) we will generate a new SIP password for this user. When re-importing
users the existing SIP password will be preserved.
What about the existing imported us
I just tested this in my install.
If i'm not logged in into my web ui, then "delete message" link really does
not work.
But when i loogged in, it works.
Rgds,
Nikolay.
_
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitc
Roman,
There is definitely not enough information to help you.
Please post a trace of this call (filtered xml suitable for sipviewer,
containing this call only).
http://wiki.sipfoundry.org/display/xecsuserV4r2/Display+SIP+message+flow+usi
ng+Sipviewer
Sipviewer shows internal interprocess message
A call trace would be helpful.
Normally I would think there is a problem with the siptrunk configuration,
but the call behavior you describe makes me think the phone device itself is
signalling in a way that needs to be scrutinized.
Does this same behavior happen with xlite?
On Fri, Oct 8, 2010
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