Hi,
I think I have a possible solution.
I was reading through some of the files and posts and there was an important
statement, not sure on the relevance, but here goes.
Following Jeff's instructions create a GoDaddy certificate in /root/sslcert (or
where ever) and run all the commands up to t
On 1/18/2010 12:14 PM, Scott Lawrence wrote:
> On Mon, 2010-01-18 at 11:31 -0600, mkitchin.pub...@gmail.com wrote:
>
>> On 1/18/2010 11:13 AM, Scott Lawrence wrote:
>>
>>> Frankly, I wouldn't invest much in getting externally generated keys
>>> working in a 4.0 system. There turn out to
*bump*
Josh Patten wrote:
That would be a good option. I'd like to hear from the developers on the
matter too.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 1/14/2010 8:14 AM, Jim Canfield wrote:
On Wed, Jan 13, 2010 at 4:07 PM, Josh Patt
Have made any changes to the ssl cert or do you have any functions/directories
on remote servers?
Sent via BlackBerry from T-Mobile
-Original Message-
From: Orrin Doyle
Date: Mon, 18 Jan 2010 16:43:38
To:
Subject: [sipx-users] TUI Change Pin Error
_
Hello all,
I'm almost done setting up a small site (maybe 100 users) but I can't seem
to get the change pin functionality from the TUI to work. Testing this
functionality results in the following:
1) Call into voicemail
2) Dial 5 for Personal Options
3) Dial 5 for Change Pin
4) Enter Current
Just fixed my issue, downgraded to version 5.25 and tried it and it
worked Aslo tried 5.28 and it also works but 5.25 seems to get
faster responses I'm downgrading all my phones to 5.25.
Thanks everyone and I hope this will help someone else too.
Jhony
On 1/18/2010 5:39 PM, Pi
I've tried all DTMF modes before, just tried in-band again and that
one doesn't work at all, if I leave it on auto or try AVT then it works
to select the extension just fine but when I dial the pin it loops back
and ask for the extension again. If I put the wrong pin it says it's the
wrong pi
Hi there,
I have a client that is looking to roll out a SipX in a hosted
configuration. They are looking for a SipX expert to ride shot gun and
provide services and development assistance. I am hoping to hear from a few
of you who feel you have strong experience with SipX that could fit this
bill.
You talk like that again, I'll hit you with my cane!
-Original Message-
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Monday, January 18, 2010 3:38 PM
To: Todd Hodgen
Cc: 'steven warner'; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Lines? Users?
You're only the se
CentOS and Red Hat are binary compatible, so you should be able to
unless Verizon has done something to their CentOS installations that
won't allow it to work.
Sven Evensen wrote:
>
> Hi guys,
>
>
>
> Has anyone run sipX (4.0.4) on a Red Hat 5 OS. I am going to try sipX
> in a Verizon CaaS
>
You're only the second person I've ever talked to that has referred to
them as "squared" lines. Must be old telephony lingo before my time. I
think you pretty much nailed it.
Todd Hodgen wrote:
> I'm sure a genius will fill in behind me. But I'll take a quick stab.
>
> The Users are your line
I'm sure everyone here loves my one sided conversations with myself :)
It was definitely the fact that the component below doesn't know how to
find my windows certificate authority, or isn't asking the way it needs
to be asked, etc. I don't expect this group to help me with that. Can
you point m
AND in users: groups is CASE sensitive:
utah is NOT Utah, is NOT Utah
On 1/18/10 5:48 PM, Michael Scheidell wrote:
GOT IT!
user 1259 in WPB group
user 1251 in Utah group
dial plan for long distance has BOTH TRUNKS
and, you can't have people in two groups.
--
Michael Scheidell, CTO
Phone: 5
Hi guys,
Has anyone run sipX (4.0.4) on a Red Hat 5 OS. I am going to try sipX in
a Verizon CaaS
cloud and they only provide Red Hat vers 4 and 5.
Sven Evensen, Technical Support Consultant
OnRelay
Elizabeth House | 39 York Road, London SE1 7NQ, UK | +44 (0) 207 902
8123 | sven.even...@on
I add |2-9]x, along with marking the group as 9 not required, and it
seems to work fine for 10 digit dialing.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jake Ballamis
Sent: Monday, January 18, 2010 2:50 PM
To: sipx-users@lis
Sorry for spreading this across multiple emails. It seems it may
definitely be the SSL certificate. I'm far from an expert in this, but
it looks like it can't figure out where to go to verify the the SSL
cert. I may have to abort and go back to the internal certificate I
guess. I don't have a c
On 1/18/10 5:49 PM, Jake Ballamis wrote:
All,
I am revisiting an old dial plan situation.
In a nutshell, my current dial plan for all phones is:
[2-9]11|0T|100|101|011xxx.T|9011xxx.T|9[2-9]x|*xx|[8]xxx|[2-7]xx
With this dial plan, I must dial a 9 to get out.
I would like to be able
On Mon, 18 Jan 2010 17:30:33 -0500, Picher, Michael wrote:
> I'd say you can run it in testing but not production... IMHO.
Might there be some way of having two then, which are kept in sync?
Or for that matter, a totally separate, non sipx vm side?
___
On Utah Gateway, you have it set for just the group that 1251 is a part of.
Example Utah Group
On WPB Gateway, you have it set for just the group that 1259 is in. Example
WPB group
Under each Gateway, you define with Dial Plan they can access - Local, LD,
Toll Free, etc.
When 1251 dials o
All,
I am revisiting an old dial plan situation.
In a nutshell, my current dial plan for all phones is:
[2-9]11|0T|100|101|011xxx.T|9011xxx.T|9[2-9]x|*xx|[8]xxx|[2-7]xx
With this dial plan, I must dial a 9 to get out.
I would like to be able to dial out without dialing the
GOT IT!
user 1259 in WPB group
user 1251 in Utah group
dial plan for long distance has BOTH TRUNKS
and, you can't have people in two groups.
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
> *| *SECNAP Network Security Corporation
* Certified SNORT Integrator
* 2008-9 Hot Company
On 1/18/10 5:37 PM, Picher, Michael wrote:
There's an issue with using location based gateways in dial plans.
Make sure you only have one user group defined for the user. If that
user is in multiple groups you won't be able to make sure you know
which group is the first one in the databas
I'm sure a genius will fill in behind me. But I'll take a quick stab.
The Users are your lines, and on a user basis, you can chose what options
are used, such as what gateway you access as just one example. If you were
a paralegal as an example, you could make calls from line 1 for attorney 1
You must use in-band dtmf...
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jhony Perez
Sent: Monday, January 18, 2010 4:16 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Linksys Phones (spa922
There's an issue with using location based gateways in dial plans. Make
sure you only have one user group defined for the user. If that user is
in multiple groups you won't be able to make sure you know which group
is the first one in the database (not the one listed first because they
are displa
Alittle more info: I looked for active logfiles when I was attempting to
make a change. httpd_access_log seems to have some relevent info. I'm at
extension/user 0142. This is me trying to change my pin from 1000 to
and it failing.
[18/Jan/2010:16:34:22 -0600] nshpbx1.sipx.voip TLSv1 DHE-RS
It only matters if you want to select voicemail for that account or if
you want to leave the other line open for calls, or if you have multiple
people sharing a phone and they want who they are calling to see the
call as coming from them.
For inbound to the phone it allows you to identify what ext
It is a dumb question to have to ask. So if anyone can point me to
the page or description of "what's what" I'd appreciate it.
Scenario:
Polycom phone, line 1 is: user 201, Line 2 is : 202. (etc)
What is the diff, if I press Line 1 and dial or
Line 2? This line vs User thing always bends me
I'd say you can run it in testing but not production... IMHO.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
m...@grounded.net
Sent: Monday, January 18, 2010 3:45 PM
To: sipx-users
Subject: [sipx-users] Stand al
I'm not sure when this started, so I don;t even want to guess as to what
might have caused it.
When an user tries to change their PIN from the voice mail menu, it
fails and they hear this:
"An error occurred while processing your request. Your personal
identification number is not changed. "
I h
Have you tried with a different brand of phone to ensure its not an issue
with the phone itself - xlite is a simple one to work with.
From: Michael Scheidell [mailto:scheid...@secnap.net]
Sent: Monday, January 18, 2010 1:22 PM
To: Todd Hodgen
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [si
yes
(would be nice for a log to trace this :-)
On 1/18/10 4:19 PM, Todd Hodgen wrote:
How about under User/Permissions -- do you have that Dial plan enabled
for the user?
*From:* sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Michael
How about under User/Permissions - do you have that Dial plan enabled for
the user?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael
Scheidell
Sent: Monday, January 18, 2010 1:03 PM
To: Dale Worley
Cc: sipx-users@list.sipfoundry.o
Hello everyone,
I have a group of Linksys phones (about 40) and after getting them to
register which took a while but I understand now the issue with the
upper case config file.
The issue is when we dial the voice mail extension we get asked to enter
the extension, once we enter the extens
On Mon, Jan 18, 2010 at 4:12 PM, Scott Lawrence wrote:
>
> Not even close.
>
> The limit is in the tens of agents - you're off by a few orders of
> magnitude.
>
>
Agent Limitation I'm aware of.. and that I understand... I'm interested in
the "Line" limitation.
___
On Mon, Jan 18, 2010 at 3:26 PM, Josh Patten wrote:
> Something tells me the current ACD engine would die on you if you did that
> to it. I don't even know that the ACD engine that will be released in 5.0
> would be able to handle that kind of load. I'll let the developers chime in
> on this one.
On Mon, 2010-01-18 at 14:17 -0500, James R wrote:
> Has anyone built a large number of lines on the ACD with success? I'm
> talking like more than a thousand.. upwards of 10,000.
Not even close.
The limit is in the tens of agents - you're off by a few orders of
magnitude.
___
its a 'dial plan' permissions thing.
under dial plan if I have: (enabled, and in this order), it works.
Long distance
International
Toll Free
Voicemail
I copy long distance to 'Utah Out' EXACTLY, and enable permissions of
"long Distance", in this order, still works:
Utah Out
Long distance
Inte
Is it possible to have two sipx servers sharing one stand alone vm server? I
basically want to run two locations with the same user accounts on each but
need to make sure that vm is always available.
There is so little information on doing these sorts of things, I've been trying
to figure out
Something tells me the current ACD engine would die on you if you did
that to it. I don't even know that the ACD engine that will be released
in 5.0 would be able to handle that kind of load. I'll let the
developers chime in on this one.
James R wrote:
> Has anyone built a large number of lines
On Mon, 2010-01-18 at 13:36 -0500, Michael Scheidell wrote:
> installed and working, see lots of pretty xml using more when I trace
> a call that went through.
>
> but, when I trace the failed calls:
>
> sipx-trace --all-components --output /tmp/sipxtrace.txt 561706@
>
> I get this:
>
> mo
Has anyone built a large number of lines on the ACD with success? I'm
talking like more than a thousand.. upwards of 10,000.
If this has been done before, what is the performance / resource impact? is
it mainly memory or CPU?
Also does anyone see any problems populating this data via postress/
On Mon, 2010-01-18 at 13:36 -0500, Michael Scheidell wrote:
>
> installed and working, see lots of pretty xml using more when I trace
> a call that went through.
>
> but, when I trace the failed calls:
>
> sipx-trace --all-components --output /tmp/sipxtrace.txt 561706@
>
> I get this:
>
>
Thanks all for insights.
I'm not sure what went wrong, but have successfully backed out of it
by simply running
/usr/bin/ssl-cert/gen-ssl-keys.sh then /usr/bin/ssl-cert/install-
cert.sh. My copy of /usr/bin/ssl-cert/gen-ssl-keys.sh still has the
2048 byte key change, and it seemed to work O
On 1/18/10 1:47 PM, Todd Hodgen wrote:
If it is a case that you want one phone to dial out on one gateway,
and the other to dial out on the other gateway, I have some ideas on that.
yep, thats what I did.
(or think I did). not that it matters, but I was programming dialplans
on mitel pbx'
No still not got it working
Simon
At 19:32 13/01/2010, mkitchin.pub...@gmail.com wrote:
>Did you have any luck resolving this? I think I have run across the
>same issue.
>
>On 1/7/2010 11:58 AM, Simon Moore wrote:
>>Hi,
>>
>>I am having trouble with getting the Auto Attendant forwarding to
>>exte
If it is a case that you want one phone to dial out on one gateway, and the
other to dial out on the other gateway, I have some ideas on that.
You need to configure the gateways under Devices/Gateways as non-shared
On the same screen, change the Location to the group you want to access the
gat
On 1/18/10 1:15 PM, Dale Worley wrote:
As with any problem like this, the best diagnostic is to run a trace of
a failed call, and see what went wrong:
http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer
Dale
installed and working, see lots of pretty xml using m
On 1/18/10 1:21 PM, Todd Hodgen wrote:
You can disable access to all gateways on an individual phone, and
then enable just the one you are working on to test it. The question
will be is it accessing the gateway and getting denied by the gateway,
or is it getting denied by sipXecs because of
You can disable access to all gateways on an individual phone, and then
enable just the one you are working on to test it. The question will be is
it accessing the gateway and getting denied by the gateway, or is it getting
denied by sipXecs because of a dial string issue, lack of permission, etc.
As with any problem like this, the best diagnostic is to run a trace of
a failed call, and see what went wrong:
http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer
Dale
___
sipx-users mailing list sipx-users@list.sipfoundry.
On Mon, 2010-01-18 at 11:31 -0600, mkitchin.pub...@gmail.com wrote:
> On 1/18/2010 11:13 AM, Scott Lawrence wrote:
> > Frankly, I wouldn't invest much in getting externally generated keys
> > working in a 4.0 system. There turn out to be many potential pitfalls,
> > and it's clear that feature rea
Iv tried lots of things, can't get everything to work I want. yes,
sorry, I know it vague, but how do I diagnose dial plan problems?
I can't even tell what gateway it thinks it wants to use. is their a
CLI log in /var/log/sipxpbx I can tail?
might be nice, in 'diagnostics' to have a dial pl
I did mine on 4.0.2 with a 1024 bit key with godaddy.
I suspect the key length and the manual changes made by someone to get it to
accept 2048 bit might moreso be the issue.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartme
On 1/18/2010 11:13 AM, Scott Lawrence wrote:
> Frankly, I wouldn't invest much in getting externally generated keys
> working in a 4.0 system. There turn out to be many potential pitfalls,
> and it's clear that feature really wasn't ready for prime time. We're
> investing a fair amount of effort
On Mon, 2010-01-18 at 11:00 -0500, Jeff Gilmore wrote:
> I tried doing it all through the web GUI) after first changing the
> underlying /usr/bin/ssl-cert/gen-ssl-keys.sh script (which I assume
> gets called by the web GUI) so that it will generate a 2048 byte key,
> since GoDaddy and other CAs are
I think you're good as long as you don't enable both.
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: 434.984.8427
Helpdesk Contra
I have it disabled on sipxbridge. It is enabled on the phones (by
default I now understand). I 'm not having any issues. I don't want to
manually change the settings on every phone if I can avoid it. I just
wanted to make sure I wasn't using a setup that was known to have issues.
On 1/18/2010
I tried doing it all through the web GUI) after first changing the underlying
/usr/bin/ssl-cert/gen-ssl-keys.sh script (which I assume gets called by the web
GUI) so that it will generate a 2048 byte key, since GoDaddy and other CAs are
now requiring longer keys.
It did seem to generate a longe
NOT WORKING! Grant has sent me a private email pointing out a problem that I
too am experiencing after my supposedly successful installation of the SSL Cert
from GoDaddy.
If you go to System/Servers and look at the list of sipx processes, they all
appear as Status: Undefined. I think that s
Yes... remove it.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Austin
Curry
Sent: Monday, January 18, 2010 10:21 AM
To: Tony Graziano
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Polycom firmware Device File Activatio
I deactivated previous firmware and removed the old .zip files from the
directory.
The firmware was downloaded direct from Polycom and uploaded in zip
format via Sipxecs
I was wondering if I should/can delete the SoundpointIPLocalization
directory, remove and upload firmware.
Austin Curry
I am going to assume from the error that at one time something was manually
uploaded or unzippped outside of sipxconfig.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Hel
You need to deactivate any firmware that is active.
You need to upload the original ZIP file from Polycom, do not unzip it.
On Mon, Jan 18, 2010 at 10:12 AM, Austin Curry wrote:
> I am having an issue when activating the polycom 3.1.3C firmware.
>
> When attempting to activate the firmware fil
I am having an issue when activating the polycom 3.1.3C firmware.
When attempting to activate the firmware file, I get redirected to a
page: "An internal error has occurred click here to continue"
I am booted out of sipxecs and back the login screen.
Viewing the source of that page reveals the f
I just keep seeing posts about how it should be off. I won't have heavy usage
until my cutover this friday, and I want to do whatever is known to work best.
-Original Message-
From: Tony Graziano
Date: Mon, 18 Jan 2010 08:26:51
To:
Cc: Picher, Michael;
Subject: Re: [sipx-users] Music
I'll just say that it has caused problems in the past having both the
polycom and sip trunk moh enabled
Mike
From: mkitchin.pub...@gmail.com [mailto:mkitchin.pub...@gmail.com]
Sent: Monday, January 18, 2010 8:22 AM
To: Picher, Michael; Tony Graziano
Cc: sipx-users@list.sipfoundry.org
S
If you do not have any problems, why would you disable it?
On Mon, Jan 18, 2010 at 8:21 AM, wrote:
> That makes sense on how it would happen. So, I literally have to go into
> every single phone and remove it? This is a known issue right, you
> absolutely have to disable MOH on polycoms, right?
That makes sense on how it would happen. So, I literally have to go into every
single phone and remove it? This is a known issue right, you absolutely have to
disable MOH on polycoms, right? I really don't care if I have MOH at all
anywhere. I just don't want any problems.
-Original Message
Fyi, I see the 'stale' problem too it seems.
-Original Message-
From: Ola Samuelson
Date: Mon, 18 Jan 2010 11:19:42
To: Tony Graziano
Cc: Ola Samuelsson;
sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] agent status on function key(snom 320)
You can not DID directly to a hunt group or ACD queue... I suggest you
setup a phantom user (user that has no phone registered to it) and then
setup call forwarding on that user to direct the call where you want it
to go.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users
I think by default it is 'blank' in the group settings but populated in
the individual phone settings. You have to remove it from each phone.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Sunday, January 1
http://track.sipfoundry.org/browse/XX-4869
The devs don't seem to be putting the effort into the existing ACD because a
new one is being developed for inclusion in 5.0.
http://track.sipfoundry.org/browse/XX-7040
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-use
Thanks.
You haven't seen the problem with "stale" acd agent sign-in?
Is there a way to use an action url to display status of agent?
2010-01-18 11:06, Tony Graziano skrev:
The only way to interact with the ACD login/logout function from the
phone is via the keypad (*86). You cannot toggle, yo
The only way to interact with the ACD login/logout function from the phone
is via the keypad (*86). You cannot toggle, you can creat two speeddials
(one for login, the other for logout).
Unless someone wants to create a plugin to use the phones microbrowser...
On Mon, Jan 18, 2010 at 4:21 AM, Ola
Hi all!
I have a client with ACD queues and from time to time is appears as if
the agents are logged out. No calls are sent
to phones but agent status in webgui says "signed in". If he sets agent
status to out and then in again - it starts to work.
Known bug?
Note: they sign in once and stay
Ditto what todd said about a phantom user for flexible AA scheduling on
inbound calls.
If firmware 5.4 was what is supported by sipx at this time, I'd certainly
head that direction, if it were me.
You need to place your outbound calls as if you were calling with a standard
handset plugged into th
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