Anyone know the URI required to setup MWI on a non-sipxecs managed endpoint?
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Good point, thanks :-)
Tony Graziano wrote:
even so, you do not want a remote user hitting the web portal during peek voice
traffic and making all of the voice traffic problematic. so it is probably a
good idea to get a handle on traffic shaping first and foremost even with a
dedicated conn
even so, you do not want a remote user hitting the web portal during peek
voice traffic and making all of the voice traffic problematic. so it is
probably a good idea to get a handle on traffic shaping first and foremost
even with a dedicated connection 4 it.
On May 11, 2011 9:16 PM, "Carl Farringt
Thanks. There will be a dedicated adsl2+ annex-m just for voice. I'll use a
reasonable router firewall - mikrotik rb750g looks worth a shot. I have good
experience of the rb450g.
Tony Graziano wrote:
only 1 network card is supported. you should employees a firewall in front and
since you on
Yep I know all about the badcaps and the theories behind it!
"Matthew Kitchin (usenet/public)" wrote:
Fyi, if you didn't know, almost every electronics company in the world used
those bad capacitors. Dell was one of the worst hit. One Asian company broke
into a rivals office and stole their c
only 1 network card is supported. you should employees a firewall in front
and since you only have a single small connection you should also enact
adequate traffic shaping on your firewall to ensure voice traffic gets
priority.
On May 11, 2011 6:44 PM, "Carl Farrington" wrote:
> Can somebody give
Fyi, if you didn't know, almost every electronics company in the world used
those bad capacitors. Dell was one of the worst hit. One Asian company broke
into a rivals office and stole their capacitor formula. The formula they stole
wasn't complete. They pass the initial test, but swell/bulge pre
Thanks, I'll have a look.
I have used pfSense before, a few times, both on PCs, and on pc-engines boards
(alix and the earlier Geode one), but I've since found the Mikrotik to be
really quite amazing in comparison both in terms of features and cost. Their
very impressive Winbox tool runs well un
On this list, you will find excellent support for use of the PFSense router.
Very affordable, traffic shaping, and several have sample configs you can
use. There is an appliance model with three ports for about $214 you can
find online.
You might want to do a search for comments on the Mikroti
Thanks Todd. That's fairly straightforward then. I will probably look to use
the existing Windows DHCP and DNS servers and share the LAN. These sites would
tend not to have proper multi-homed routers, only simple ADSL wireless routers,
but I may add a Mikrotik router into the mix for some finer
No, you just want one nic on sipXecs. Users will connect to the User portal
based on its domain name and its port number -
yoursixpecs.local:8443/sipxconfig/apps
Treat sipXecs like any other server, with IP traffic delivered to it based
on SRV records. If you are using the sipXecs for your D
Me either. With 2.14, I got it to the point where it answered every other page.
I found others with the exact same problem. I called Valcom, and within 60
seconds, they had sent me 2.17 and all was perfect. I have it in a paging group
along with several polycom phones. It works perfect. Mine I'd
Can somebody give me some recommendations on NIC configuration on a small (15
user) deployment with a dedicated DSL line for the ITSP trunk? My background is
really in Asterisk & FreePBX, and I have to date configured those systems with
a dedicated WAN NIC, and either a separate LAN for the hand
Ahh maybe that's the hold up then it only has 2.14, but I didn't see anything
newer on their website.
Aaron Pursell, Sr.
Network Systems Administrator
Easter Seals - Goodwill, Northern Rocky Mountain
4400 Central Ave
Great Falls, Montana 59405
(406) 771-3721
aar...@esgw.org
>>> "Matthew Kitc
My Valcom works fine in a paging group. It didn't until I called them and got
firmware 2.17.
-Original Message-
From: "Aaron Pursell"
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Wed, 11 May 2011 15:19:06
To: Discussion list for users of sipXecs
software
Reply-To: Discussion li
they say the same thing in every version...
while it works, it does not work exactly as described. when they force an
update to you, it does not help if they break things leaving you pandering
to them to go back.
On May 11, 2011 6:18 PM, "Aaron Pursell" wrote:
> The people at counterpath told me
The people at counterpath told me Bria does but they said if you don't sign the
NDA to talk to them (even basic sales talk), you can purchase "test" clients at
their webpage at the tune of 50/client..
y Graziano 5/11/2011 4:11 PM >>>
I have not tested any that are both sip and x m p p that
I have not tested any that are both sip and x m p p that work worth a hoot
that also do video. to that end i nailed up my own om conference serveris
just for video. webcam and flash and I just use the built inthe audio
conference serveris in sipx. works like a champ.
On May 11, 2011 5:45 PM, "Aaron
I am actually in the process of making a stencil for the Mediant 1000.
Majority of the stencil is finished (in omnigraffle), however I still need to
complete a few RJ Jacks and attach them to modules. I could make it a multiple
part stencil so that the modules are available and can be drag and
SipXecs paging Feature - Analog Gateway FXS --- Valcom paging
adapter ---amplifier --- speakers.
Works like a champ for one way paging.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Aaron Pursell
Sent: Wednesday
Anyone tested XMPP clients with 4.4.0? We're looking for a replacement our
conferencing so it needs to support video and if we can find the right working
setup, we'll go with it.
Also we're not interested in Bria due to the fact that even to talk to them you
have to sign a 3 year NDA. So far I
I prefer any true paging device that is not as messy as using.v phones in
any true volume environment. over xx. umber of phones and you begin to have
problems or an excessive amount of lan traffic, not to mention the copious
cdrom records...
On May 11, 2011 5:19 PM, "Aaron Pursell" wrote:
> Ahh th
Ahh then it works perfectly! Still will mess with cyberdata to see if I can get
it to work, because it does support that.
Thanks for the added info
-Aaron
Aaron Pursell, Sr.
Network Systems Administrator
Easter Seals - Goodwill, Northern Rocky Mountain
4400 Central Ave
Great Falls, Montana
um... they are not designed to use the alert-info header. they register and
you use them like an extension. Dial the extension. talk. hangup.
The don't use the alert-info header because they are not 2 way devices.
Valcom has always been that way. If you have v 2.15 (I think) or later, it
works fi
It is and registers fine, but it doesn't work right in a paging group, matter
of fact doesn't work at all in a paging group.
Aaron Pursell, Sr.
Network Systems Administrator
Easter Seals - Goodwill, Northern Rocky Mountain
4400 Central Ave
Great Falls, Montana 59405
(406) 771-3721
aar...@esgw
make sure your firmware is up to date with the valcom, then it works fine.
On Wed, May 11, 2011 at 5:03 PM, Aaron Pursell wrote:
> Couldn't tell you didn't get any other information on this issue so moved
> on and put in a valcom we had sitting around which doesn't work right with
> 4.4.0 but i
Couldn't tell you didn't get any other information on this issue so moved on
and put in a valcom we had sitting around which doesn't work right with 4.4.0
but its better than the alternative. I will bring the cyberdata to my desk and
let you know.
Aaron Pursell, Sr.
Network Systems Administra
Aaron,
This could be possibly due to a bug with the UA from-Tag changing when
authenticating against sipX. Assuming the phone gets a 401 back, what are the
errors in /var/log/sipxpbx/sipregistrar.log?
Mike
Aaron Pursell esgw.org> writes:
___
sipx-
This is great Josh. Thanks!
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh M. Patten
Sent: Wednesday, May 11, 2011 11:47 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Audiocodes backup script
My gift to those man
My gift to those managing Audiocodes gateways manually:
http://wiki.sipfoundry.org/display/sipXecs/Audiocodes+Gateways+Backup+Script
Enjoy
Josh Patten
Brazos County Network Engineer
979.361.4676
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On Wed, May 11, 2011 at 11:26 AM, Tony Graziano
wrote:
> I hear it I think about that upgrading from unstable versions is supported.
>
> well I am making my own point and maybe only to myself, upgrading from
> unstable versions is not implicitly supported until these issues are
> resolved.
Upgrad
On Wed, May 11, 2011 at 11:27 AM, Matthew Kitchin (public/usenet)
wrote:
> On 5/10/2011 11:24 PM, Douglas Hubler wrote:
>> This is a notice in accordance with bug fix release policy
>>
>> http://wiki.sipfoundry.org/display/sipXecs/Bug+Fix+Release+Policy
>>
> If it was determined one of the bug
NetZoom 11. Expensive but has darn near every device ever created... :)
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew
Kitchin (public/usenet)
Sent: Wednesday, May 11, 2011 11:25 AM
To: Discussion list fo
I use 3.2.4 and 3.2.5 without issue on current models.
On Wed, May 11, 2011 at 1:17 PM, Morgan, Michael wrote:
> All,
> I have been away from SIP Foundry for a while. What version of Polycom
> firmware should I have with 4.4.0. I know we were limited by 3.2.1 I think?
>
> -Michael Morgan
>
>
All,
I have been away from SIP Foundry for a while. What version of Polycom
firmware should I have with 4.4.0. I know we were limited by 3.2.1 I think?
-Michael Morgan
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Hi,
Without knowing the exact specs of those machines, I'd choose the one
with bigger L2(/L3) cache and/or FSB since sipx has java processes and
operates on large datasets.
We found, that our HP DL360 with E5405(2.0GHz, 12MB L2, 1333MHz FSB) was
considerably faster, than our DL320 with X3210(2.13
global is the publicly advertised IP address. All those settings are
correct.
what you should be comparing is the itsp account settings for the skype
account. I would think it would say "use-global-addressing=true"
you ought to create two skype entries (gateways) with different names, to
compare
Okay, I've started looking at the sipxbridge.xml and it looks like the
"Use public address..." setting is using the "external-address" instead
of the global-address. Is the external address supposed to be my public
IP or my internal IP?
70.xxx.xxx.xxx
192.168.81.23
5080
192
All,
We have fixed this by adding two trunks for Bandtel, one with proxy1 and
another with proxy2 and added these trunks to the dial plan. Now we can receive
call from both proxies.
Thanks,
Sushil
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfo
On 5/10/2011 11:24 PM, Douglas Hubler wrote:
> This is a notice in accordance with bug fix release policy
>
> http://wiki.sipfoundry.org/display/sipXecs/Bug+Fix+Release+Policy
>
If it was determined one of the bugfixes caused an issue, is there a way
to back it out?
> Notes
> =
it may not matter, this was also seen updating to 4.2.1 by someone else. in
both of these instances, they were coming from unstable versions.
I hear it I think about that upgrading from unstable versions is supported.
well I am making my own point and maybe only to myself, upgrading from
unstable
Has anyone ever found any Visio stencils for Audiocodes products?
Specifically the MP 114? I can't find them anywhere.
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Hi,
Ah thanks for the reply, that would be my problem I guess...
I googled and found this:
https://supportforums.cisco.com/message/3027896#3027896
So they are trying I guess but failing J
Let's hope this gets fixed some time.
Henry
From: Jean-Hugues Royer [mailto:jhro...@joher.com]
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Irena Dolovčak
[irena.dolov...@gmail.com]
our logs show that this is happening: call comes from the itsp's server to the
bridge on sipx1. sipxbridge
forwar
Hi,
To be able to see the presence of the phones, the phones need to support
subscription for event type dialog (RFC4235).
All the Linksys products SPA941/942/962 do _not_ support it.
Since SPA905G is a revised SPA942 there is fair chance that it doesn't
support it either.
But you should c
Timeout for PIN is not a bad idea. But I think we can take that as
phase 2 of this issue.
Because it will not be a simple addon to this.
_lvnd_
(^_^)
On Wed, May 11, 2011 at 3:24 PM, Michael Picher wrote:
> Hi Levend,
>
> Not sure this is the right project to capture this under but sometimes t
By the way sorry for calling you John J I meant Jean..
Henry
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Henry
Dogger
Sent: woensdag 11 mei 2011 15:27
To: Discussion list for users of sipXecs software
Cc: jhro...@joher.com
Subjec
On Wed, May 11, 2011 at 10:17 AM, Dave Deutschman
wrote:
> +1
http://track.sipfoundry.org/browse/XX-9609
FYI: It would be very hard to base this on context of bug fixes.
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On Wed, May 11, 2011 at 9:26 AM, Tony Graziano
wrote:
> I had planned that. What I didn't understand is why none of the logging
> components failed to capture the issue and why there was a reported
> "ConfigurationMismatch".
> While everything reported properly in services, the fact that the
> Pre
Hi Users,
I finally got the green light to start ramping up our SIPX install on
main campus and I'm looking into upgrading our servers to support 8000
users.
I'm looking at basically two options and AMD based machine and an
Intel based machine.
AMD:
4 processors (2 GHz), 8 cores each = 32 cores!
+1
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitchin
(public/usenet)
Sent: Wednesday, May 11, 2011 6:26 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Bug fix release
We are using Bandtel as our trunk provider and have used the Bandtel
template to setup the SIP trunk. Bandtel has two proxies, proxy1 and
proxy2 . We are experiencing a problem where the incoming calls sent
via Prox1 works correctly and the calls sent via. Proxy2 does not
work. We traced the calls
I had planned that. What I didn't understand is why none of the logging
components failed to capture the issue and why there was a reported
"ConfigurationMismatch".
While everything reported properly in services, the fact that the
PresenceServer had a log of "0" bytes make me think it was not runn
Hi John,
I downloaded the trial version, and successfully configured it using the
tutorial on the website (thanks for that)
But do you know/encountered any issues with cisco spa504G phones?
I am able to see the presence of my polycom phones, but not from the
cisco's...
Any help or advice wou
On Wed, May 11, 2011 at 8:17 AM, Gerald Drouillard
wrote:
> On 5/11/2011 4:29 AM, barisyanar wrote:
> > Raised more questions:
> >
> > 1 - IP Ban: How would this work if the call is made via gateway (e.g.
> > Audiocodes). Should we do something with caller ID?
> Just like in fail2ban you can choos
On 5/11/2011 7:49 AM, Douglas Hubler wrote:
> On Wed, May 11, 2011 at 7:44 AM, Tony Graziano
> wrote:
>> Please note: this will repush profiles to end user devices as well as reboot
>> them, so shedule/apply accordingly.
> Would everyone agree we should change this so it only happens on major
> v
Correct.
On Wed, May 11, 2011 at 9:08 AM, Tony Graziano wrote:
> template "or the resulting config file"...
>
> meaning if incorrect values are corrected by result of the patch without an
> actual change to the template, then of course the phones need the new
> config...
>
>
> On Wed, May 11, 20
template "or the resulting config file"...
meaning if incorrect values are corrected by result of the patch without an
actual change to the template, then of course the phones need the new
config...
On Wed, May 11, 2011 at 8:59 AM, Michael Picher wrote:
> That would make sense to me.
>
> I gues
That would make sense to me.
I guess we just need to note in the release notes if there's a phone
template change that might require that the admin re-publish profiles.
Mike
On Wed, May 11, 2011 at 8:49 AM, Douglas Hubler wrote:
> On Wed, May 11, 2011 at 7:44 AM, Tony Graziano
> wrote:
> > Pl
On Wed, May 11, 2011 at 7:44 AM, Tony Graziano
wrote:
> Please note: this will repush profiles to end user devices as well as reboot
> them, so shedule/apply accordingly.
Would everyone agree we should change this so it only happens on major
version upgrades?
_
See inline...
On Wed, May 11, 2011 at 8:17 AM, Gerald Drouillard
wrote:
> On 5/11/2011 4:29 AM, barisyanar wrote:
> > Raised more questions:
> >
> > 1 - IP Ban: How would this work if the call is made via gateway (e.g.
> > Audiocodes). Should we do something with caller ID?
> Just like in fail2ba
Hi Levend,
Not sure this is the right project to capture this under but sometimes there
is a request for timeout of PIN. For instance, a PIN might only be valid
for XX days and require a change. The change must not be the same as the
previous PIN (or X pins, where X might be 1 - 5).
Thanks,
M
On 5/11/2011 4:29 AM, barisyanar wrote:
> Raised more questions:
>
> 1 - IP Ban: How would this work if the call is made via gateway (e.g.
> Audiocodes). Should we do something with caller ID?
Just like in fail2ban you can choose to whitelist IP's or networks.
>
> 2- Account blocking: Shouldn't th
Please note: this will repush profiles to end user devices as well as reboot
them, so shedule/apply accordingly.
On Wed, May 11, 2011 at 12:24 AM, Douglas Hubler wrote:
> This is a notice in accordance with bug fix release policy
>
> http://wiki.sipfoundry.org/display/sipXecs/Bug+Fix+Release+P
On Tue, May 10, 2011 at 9:15 AM, Michael Picher wrote:
> Issues like this are why I use an external session border controller...
> specially with HA systems.
>
> Mike
>
> On Tue, May 10, 2011 at 6:38 AM, Irena Dolovčak <
> irena.dolov...@gmail.com> wrote:
>
>> Hi guys,
>>
>> we have a problem
Weren't we supposed to get descriptions on those fields?
I think it means call will ring until agent is available. That is, folks
don't get put on hold. Yet another way to think about it: kinda like a be
hunt group.
This is how it was explained to me, I never tried it though.
On May 11, 2011 4:47
On Tue, May 10, 2011 at 7:39 PM, Aaron Pursell wrote:
> Right, it doesn't work if your outside this SIP server however it resolves
> to the last "at same time user" when our HQ users call in. They are bringing
> it up as a concern because while they think they're calling the Hunt Group
> (and they
Hi Dale,
we are not sure what you mean by 'The trunk should be given the SIP domain
name as the destination for calls'.
the only thing that we can think that this might apply is
DEVICES->Gateways->NAME_OF_GATEWAY->Configuration->ADDRESS,
and there we have the ip address that our itsp gave us. also
Improvement raised as XX-9606
2011/5/11 George Niculae
> On Wed, May 11, 2011 at 11:56 AM, Anders Mydland
> wrote:
> > George,
> >
> > Multi-value attributes from LDAP _do_ work for aliases, but are not very
> > practical in a Windows / Active Directory environment.
> > Windows user profile ty
On Wed, May 11, 2011 at 11:39 AM, Levend Sayar wrote:
> Hi everybody,
>
> I want to add this feauture. This issue has two sides, one is VM, the
> other is user portal.
>
> My ideas :
>
> 1) Add a field (PIN change required flag) to user db, set default as
> "required"
> 2) Add a checkbox for this
On Wed, May 11, 2011 at 11:56 AM, Anders Mydland wrote:
> George,
>
> Multi-value attributes from LDAP _do_ work for aliases, but are not very
> practical in a Windows / Active Directory environment.
> Windows user profile typically have predefined attributes such as
> sAMAccountName, ipPhone, bus
Hello,
I have find in the documentation where are the localized template, under
/etc/sipxpbx/region_xx
I want to change the region_ch template.
How can I add a custom rule ?
Where can I find the list of the properties to use in the xml custom dial plan
?
Regards
Pierre Dubey
_
George,
Multi-value attributes from LDAP _do_ work for aliases, but are not very
practical in a Windows / Active Directory environment.
Windows user profile typically have predefined attributes such as
sAMAccountName, ipPhone, businessPhone and the like that could all go into
"aliases".
On a side
On Wed, May 4, 2011 at 11:28 AM, Kumaran <
thiru.venkateshwa...@ttplservices.com> wrote:
> Hi All,
> I'm testing OpenACD on the build "sipXconfig (0.0.4.5.2-
> 2011-04-26EDT08:49:49 domU-12-31-39-05-28-E2)".When Adding a line there
> is option "supervision check box" which is enabled by default.
Hi everybody,
I want to add this feauture. This issue has two sides, one is VM, the
other is user portal.
My ideas :
1) Add a field (PIN change required flag) to user db, set default as "required"
2) Add a checkbox for this field to user page in admin portal
3) Redirect user to a new page on log
Raised more questions:
1 - IP Ban: How would this work if the call is made via gateway (e.g.
Audiocodes). Should we do something with caller ID?
2- Account blocking: Shouldn't this be a generic mechanism including also
user portal access failures and then password renewal automation etc. (Does
th
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