I think I know why the files didn't restore. They didn't backup either.
I just took a current backup, and checked the files. My newly imported
wave files didn't make the backup. It is because they had spaces in the
names. I just created 2 test files in
/var/sipxdata/configserver/phone/acd/audio
I found a completely unrelated issue, but they were having a file upload
problem as well.
http://track.sipfoundry.org/browse/XX-1717
(search for RequestExceptionReporter to find the relevant part)
I thought it was a permission issue for sure, but I did not want to mess
anything up. I looked at a
On Wed, Jan 20, 2010 at 4:03 PM, Michael Scheidell wrote:
> I toyed around with a couple of them, one of them that wasn't on the
> 'certified' list, but defaults seemed to work (ipcomms).
There is no 'certified' list - only a list which catalogs successful
interoperability by users. Please suppl
You knew this one was coming...
After I finished my restore, some things weren't behaving correctly.
First, the custom wav files uploaded for auto attendants were gone. Not
a huge deal. I added them back, and that was fine.
Second, custom wav files for ACD Queue were gone. The drop down box just
Hi Charles,
That's an interesting problem you have there.
If you can get some cooperation at the other end, it might be worthwhile to
schedule a caller (one that is known not to work properly) and not only
supply a sip trace, but also do a good patton debug file and ask patton to
look at it as we
I am running sipx 4.0.4 with 4 pstn lines through a patton smartnode 4524 to my
sip box with polycom 650 or 601s. I have a few inbound calls where we see the
phones ringing with the caller id but are unable to pick up to answer the
phone. This probably happens only a few times a week but inter
My pain from today can confirm this issue.
When I restored my sipx config, firmware updates were broken. The files
were still checked in, and everything looked fine, but a new handset
wouldn't update. I deleted the device files, re-uploaded the exact same
files, and everything was fine.
On 1/6/
Tried this experiment on second phone: Set lineKeys to 3. (the group
setting is 2). Then set it back to blank. Now the group setting has no
effect. Since the phone's setting takes precedence over the group it's
as if it interprets even an empty value as a "setting" and this
overrides the group valu
Yes, it's in the one and only group. Still doesn't work.
On Wed 20.Jan.10 14:48, Michael Scheidell wrote:
>Try making sure you are in only ONE group, and its CasE sensitive.
>
>
>--
>Michael Scheidell, CTO
>Phone: 561-999-5000, x 1259
>> *| *SECNAP Network Security Corporation
>
> * Certified S
I toyed around with a couple of them, one of them that wasn't on the
'certified' list, but defaults seemed to work (ipcomms).
I have skype beta, and now I am ready to look at sip trunk
recommendations, mostly for FX (remote DID's).
We will be having 23 SIP trunks coming in from Level3, so main
FYI, Pathlen is 0 on my box I did DR testing to a few weeks ago.
On 1/20/2010 2:40 PM, Raymond Dans wrote:
> mkitchin.pub...@gmail.com wrote:
>
>> Subject: Re: [sipx-users] SSL Cert help
>>
>> Am I dead in the water here? Should I go ahead and reload the OS?
>>
> I'm fresh out of ideas.
That should be handled in the [2-9]xx.T
Mike
From: Jim Canfield [mailto:jcanfi...@emstar.com]
Sent: Tuesday, January 19, 2010 10:07 PM
To: Picher, Michael
Cc: Jake Ballamis; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Dial plan questions
On Tue, Jan 19, 2010 at 4:18 AM,
Get a pfSense box in place... you don't need much hardware and it's a
known good combination.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Eric
Varsanyi
Sent: Tuesday, January 19, 2010 12:18 PM
To: Scott
The fact that Phone settings "invisibly" override Phone Group settings
has me tearing out my hair. It's the single issue with sipXecs that
causes me the most grief.
I have few enough phones that it's not TOO much hassle to document which
phone belongs to which user, delete all of them, and re-dis
Date is ok.
[r...@nshpbx1 ssl]# date
Wed Jan 20 14:42:53 CST 2010
I'm sure I did something to cause this. Unless I hear something I will
start reloading the OS in about 20 min.
On 1/20/2010 2:40 PM, Raymond Dans wrote:
> mkitchin.pub...@gmail.com wrote:
>
>> Subject: Re: [sipx-users] SSL Cer
mkitchin.pub...@gmail.com wrote:
>Subject: Re: [sipx-users] SSL Cert help
>
>Am I dead in the water here? Should I go ahead and reload the OS?
I'm fresh out of ideas.
The one thing that seemed odd to me in your output from keytool was that
the PathLen in both show as undefined under BasicConstr
Am I dead in the water here? Should I go ahead and reload the OS?
On 1/20/2010 1:34 PM, mkitchin.pub...@gmail.com wrote:
> On 1/20/2010 1:30 PM, Raymond Dans wrote:
>> I believe it looks okay. The beginning of that line is for the alias
>> name and you don't have one. I'm not familiar enough wit
Thanks. I had used those in the past and recovered fine. Somewhere in
the process of trying to make the internal MS cert work, I must have
caused an issue of some sort. The MS certs didn't have the right name,
extension, etc, so I had to do a few more things.
On 1/20/2010 1:47 PM, Grant Lang wr
Try making sure you are in only ONE group, and its CasE sensitive.
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
> *| *SECNAP Network Security Corporation
* Certified SNORT Integrator
* 2008-9 Hot Company Award Winner, World Executive Alliance
* Five-Star Partner Program 2009,
Hi,
This page here has probably most of the information required to get you back to
a 'known' state, but basically (as root) run:
mkdir $HOME/sslkeys
cd $HOME/sslkeys
/usr/bin/ssl-cert/gen-ssl-keys.sh
Then:
/usr/bin/ssl-cert/install-cert.sh
http://sipx-wiki.calivia.com/index.php/SSL_Certific
On Wed, 2010-01-20 at 20:18 +0300, Nikolay Kondratyev wrote:
> Hi all,
>
> I use Russian display names with snom360 phones. Sipx 4.0.4.
>
> When the call is between two users of the same sipx all working fine,
> one can see Russian name of the caller.
>
> But when the call is between two sipx sy
On 1/20/2010 1:30 PM, Raymond Dans wrote:
> I believe it looks okay. The beginning of that line is for the alias
> name and you don't have one. I'm not familiar enough with this to know
> whether have no alias is okay.
>
> Try issuing the keytool commands again and add '-v' after the '-list'.
> T
On Wed, 2010-01-20 at 14:27 -0500, Dale Worley wrote:
> On Wed, 2010-01-20 at 14:05 -0500, Scott Lawrence wrote:
> > Actually, I think that's an illusion. I've looked more closely and
> > repeated the unit test of the C++ Url class and it seems to work.
> >
> > I think the illusion is created whe
>Subject: Re: [sipx-users] SSL Cert help
>
>Everything looks ok to me, I think. The only possibly odd
>thing I see is the 'trustedCertEntry' part in the keystore. I
>did a disaster recover test of this machine a few weeks ago
>and I looked at the machine I recovered it to then, and it has
>the
On Wed, 2010-01-20 at 14:05 -0500, Scott Lawrence wrote:
> Actually, I think that's an illusion. I've looked more closely and
> repeated the unit test of the C++ Url class and it seems to work.
>
> I think the illusion is created when the logs are merged - the first
> message that shows as incorr
Everything looks ok to me, I think. The only possibly odd thing I see is
the 'trustedCertEntry' part in the keystore. I did a disaster recover
test of this machine a few weeks ago and I looked at the machine I
recovered it to then, and it has the full name of the root cert at the
beginning of t
I can't get one of my SoundPoint 501s to accept a setting from the
group. I had set up multiple line appearances on one phone. After
testing I removed them from the phone and added them to the group. The
second 501 picked them up, but the original one won't.
I've tried removing it from the group,
I filed an issue http://track.sipfoundry.org/browse/XX-7460
And I found a strange thing that wireshark shows that the problem is in
different place... (described in the issue).
Thanks and regards,
Nikolay.
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-us
>Subject: Re: [sipx-users] SSL Cert help
>
>On 1/20/2010 12:27 PM, Raymond Dans wrote:
>> Not sure if this will help but did you regenerate and
>install the Java
>> Keystore/Truststore? If not you may want to try this first.
>I'm not sure exactly how to do that, so I guess I hadn't. How
>should
On Wed, 2010-01-20 at 12:45 -0500, Dale Worley wrote:
> On Wed, 2010-01-20 at 12:42 -0500, M. Ranganathan wrote:
> > I would suspect that it is a sipxconfig limitation. The From header
> > display name is obtained from the dial plan and is presented to
> > sipxbridge in the INVITE. SipXbridge simp
I'm not sure exactly how to do that, so I guess I hadn't. How should I
do that? The ssl script seems to indicate it is doing that (see below).
On a side note, I just tried completely rerunning the sipx setup wizard.
That didn't help. Same result.
I realize my timing here is awful. I am desperate.
>Subject: Re: [sipx-users] SSL Cert help
>
>I will be glad to listen to a whole bunch of "I told so", but
>I would greatly appreciate a little help first.
>I made a system backup, and backed up the SSL directories
>before trying any of this. I wanted to give an external SSL
>cert one more shot.
On Wed, 20 Jan 2010 13:18:13 -0500, Tony Graziano wrote:
> I think the thing you need to do is to post some of the questions of the
> sipx-dev list, because the voicemail server is being replaced in its
> entirety in 4.2 with freeswitch.
I thought about that but not being a programmer, I didn't th
Possible with our ACD
http://wiki.voiceworks.pl/display/vwost/VoiceWorks+ACD
one production site has 100 agents online, talking. Line/queue number is
unlimited as all requests are passed to SQL.
Lookup time is depended on PostgreSQL speed with btree index so 1 should be
no problem.
No id
On Wed, 2010-01-20 at 13:02 -0500, Scott Lawrence wrote:
> If you could provide the hex or octal byte sequence for that
> name, it would help (in case I got the transcription wrong).
The .xml file he included seems to contain the proper byte sequences for
the display names (both Russian and damage
I think the thing you need to do is to post some of the questions of the
sipx-dev list, because the voicemail server is being replaced in its
entirety in 4.2 with freeswitch.
On Wed, Jan 20, 2010 at 12:20 PM, m...@grounded.net wrote:
> I've offered to spend time on this, just need some help.
>
You can also do a call forward "at the same time" from the ui for the user
and put the phone on DND. This will keep the phone from ringing, but it will
show the missed call in the phone logging.
There are 3 ways, generally
DND (see above)
Call Diversion (see jim canfields response)
The third way
On Wed, 2010-01-20 at 12:43 -0500, Dale Worley wrote:
> On Wed, 2010-01-20 at 20:18 +0300, Nikolay Kondratyev wrote:
> > I’m not sure if all of you will be able to see Russian letters, but I
> > can see that frame 10 and previous contain correct name in From
> > header, while frame 11 and subsequen
This can be done with both Asterisk and FreeSWITCH too. I currently use
my Audiocodes to deal with unwanted numbers and send them to a number
not in service prompt on my asterisk box.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 1/20/2010 11:52 AM, Micha
On 1/20/10 9:48 AM, Tony Graziano wrote:
> Yes, it would be nice to handle this internally in some fashion..
>
> When caller matches (blacklist), then forward to FCC complaint department...
>
broadsoft has two things they can do, a 'selective call fwd' (with a
couple boxes like []unknown []ano
On Wed, 2010-01-20 at 12:42 -0500, M. Ranganathan wrote:
> I would suspect that it is a sipxconfig limitation. The From header
> display name is obtained from the dial plan and is presented to
> sipxbridge in the INVITE. SipXbridge simply forwards that INVITE.
> Looking at it through wireshark, do
On Wed, 2010-01-20 at 20:18 +0300, Nikolay Kondratyev wrote:
> I’m not sure if all of you will be able to see Russian letters, but I
> can see that frame 10 and previous contain correct name in From
> header, while frame 11 and subsequent contain broken name in the From
> header.
Yes, compare fram
On Wed, Jan 20, 2010 at 12:18 PM, Nikolay Kondratyev wrote:
> Hi all,
>
> I use Russian display names with snom360 phones. Sipx 4.0.4.
>
> When the call is between two users of the same sipx all working fine, one
> can see Russian name of the caller.
>
> But when the call is between two sipx syste
I will be glad to listen to a whole bunch of "I told so", but I would
greatly appreciate a little help first.
I made a system backup, and backed up the SSL directories before trying
any of this. I wanted to give an external SSL cert one more shot. It
didn't work, so I went to revert back to a se
On Wed, Jan 20, 2010 at 11:08 AM, Rene Pankratz
wrote:
> Hello list members,
> I got a very simple question. I want to forward a call to an extension
> directly so that the called extension does not ring.
>
At this point, if you don't want the main extension to ring, using
line diversion seems to
I've offered to spend time on this, just need some help.
If it sounds too hard to do this, then I've also asked several times for
thoughts on how else I might go about creating a multi-site redundant setup but
the threads always lead to nothing.
I've been told that my questions can be 'outside
Hi all,
I use Russian display names with snom360 phones. Sipx 4.0.4.
When the call is between two users of the same sipx all working fine, one
can see Russian name of the caller.
But when the call is between two sipx systems the name is broken, it looks
like wrong encoding is used (instead of ut
Hello list members,
I got a very simple question. I want to forward a call to an extension
directly so that the called extension does not ring.
I know I can reach this when activating DND on telephone and configuring a
call forward in SipX user interface. But would'nt it make sense if I could
conf
Thanks Scott and Robert for the detailed descriptions here. I appreciate
it.
-Original Message-
From: Robert Joly [mailto:rj...@avaya.com]
Sent: Wednesday, January 20, 2010 6:19 AM
To: Scott Lawrence; Todd Hodgen
Cc: sipx-users; Peter Fowler
Subject: RE: [sipx-users] DND Presence and Int
Yes, it would be nice to handle this internally in some fashion..
When caller matches (blacklist), then forward to FCC complaint department...
Ha ha.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Se
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> Lawrence, Scott AVAYA (BL60:9D30)
> Sent: Wednesday, January 20, 2010 8:44 AM
> To: Todd Hodgen
> Cc: 'sipx-users'; Fowler, Peter AVAYA (CAR:9D10)
> Subj
On Wed, 2010-01-20 at 00:05 -0800, Todd Hodgen wrote:
>
> Has anyone had a creative workaround that allows DND to keep ALL calls
> from coming to a station – Polycom in this instance.
DND is purely local to the phone - the SIP routing elements are not
aware of it and just send any call regardless
On Wed, 2010-01-20 at 12:21 +, mkitchin.pub...@gmail.com wrote:
> Scott - if there are issues, should they show up immediately? If you
> have to back out, is it still just as easy as regenerating the self
> signed cert?
Yes, they should show up as soon as you restart.
If you think regeneratin
Scott - if there are issues, should they show up immediately? If you have to
back out, is it still just as easy as regenerating the self signed cert?
Sent via BlackBerry from T-Mobile
-Original Message-
From: Grant Lang
Date: Wed, 20 Jan 2010 07:42:07
To: 'mkitchin.pub...@gmail.com';
s
Trying to get Intercom calls to not be placed to users instruments that are
set for DND. Suspect there isn't a way to do it.
Curious if anyone has a creative workaround.DND is not seen by presence,
at least not on the multiple PC consoles I've tried. When a person turns on
DND, the ring t
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