Hi Adam,
I have attached a new patch that disables a particular check for the
presence of the qop paramater in the sipXproxy code in
http://track.sipfoundry.org/browse/XX-7997.
I have check the digest verification functions of sipx and the
implementation does allow qop to be absent. However
HI
I know that must be a topic that come back often but I did not found
anything that could help me with my problem.
At the moment I am doing the telephony system for an enterprise and they
want remote user (softphone and android user) to be able to connect the
telephony server from home or in
i was wondering if anyone knows how to set the default text input on the
minibrowser to Numbers and not lower or uppercase letters... we are
creating a speed dial application which we will be releasing to the public
once finished...
problem is the default text box input for the polycom
I wish I knew, it plagues us to this day. We have a time clock app
written for the Polycom phones so users can clock in and out and it
writes to our time and accounting database.
Aaron Pursell
Network Systems Administration
Easter Seals-Goodwill Northern Rocky Mountain, Inc.
4400 Central Ave
i wouldn't mind just turning off Alpha input off all together... we only
need Numeric input
On Fri, Mar 23, 2012 at 9:29 AM, Aaron Pursell aar...@esgw.org wrote:
I wish I knew, it plagues us to this day. We have a time clock app
written for the Polycom phones so users can clock in and out
Doesn't Polycom publish this in their xhtml developer guide? Since we are
trying to control the input on the phone, has anyone looked there?
On Fri, Mar 23, 2012 at 9:43 AM, Chris Rawlings cm.rawli...@gmail.comwrote:
i wouldn't mind just turning off Alpha input off all together... we only
need
+1 Audiocodes... mediant 1000 or 2000.From my experience, they're the
only ones if you're ever to consider t.38
On Mar 23, 2012 9:40 AM, Aaron Pursell aar...@esgw.org wrote:
We currently have 9 Mediant 1000s with T1/PRI cards and they have worked
very well for us. The only drawback with
I tried to find it as well as one of our developers, may have been
overlooked.
Aaron Pursell
Network Systems Administration
Easter Seals-Goodwill Northern Rocky Mountain, Inc.
4400 Central Ave
Great Falls, Montana 59405
(406) 771-3721
aar...@esgw.org
Tony Graziano
Our PRI connects to a Patton 4960. The biggest downside is that sipXecs
doesn't give you a starter config - however, once you get a basic configuration
from an existing install, adding additional capabilities goes very smooth.
They also offer free tech support - and their tech support is
We run t.38 on the Patton as well.
+1 Audiocodes... mediant 1000 or 2000.From my experience, they're the
only ones if you're ever to consider t.38
This message and any files transmitted with it are intended only for the
individual(s) or entity named. If
I'm assuming that javascript isn't supported
This message and any files transmitted with it are intended only for the
individual(s) or entity named. If you are not the intended individual(s) or
entity named you are hereby notified that any disclosure, copying,
Sounds like an enhancement request (to polycom). Alternately your app might
be configurable to use numbers only, and could tanspose ABC to input
:1. It sounds like a cludge, but it may work.
On Fri, Mar 23, 2012 at 10:39 AM, Chris Rawlings cm.rawli...@gmail.comwrote:
i have but didn't see
We use t.38 with Patton all the time. Have for 2 years while testing and
troubleshooting FS fax for sipx.
On Fri, Mar 23, 2012 at 10:46 AM, Positively Optimistic
positivelyoptimis...@gmail.com wrote:
+1 Audiocodes... mediant 1000 or 2000.From my experience, they're
the only ones if
Not with 4.4...
You may want to VPN anyway because many (Verizon) block sip...
Mike
On Friday, March 23, 2012, Simon Brûlé sbr...@360-innovations.com wrote:
HI
I know that must be a topic that come back often but I did not found
anything that could help me with my problem.
At the moment I
Might have to get a Patton in here, we use t.38 so its a requirement.
Aaron Pursell
Network Systems Administration
Easter Seals-Goodwill Northern Rocky Mountain, Inc.
4400 Central Ave
Great Falls, Montana 59405
(406) 771-3721
aar...@esgw.org
Nathaniel Watkins nwatk...@garrettcounty.org
Thought about that as well, only A, B and C should transfer to 2 to make
it understandablestill a cludge maybe.
Paul
Tony Graziano tgrazi...@myitdepartment.net wrote on 23-03-2012 16:19:03:
Sounds like an enhancement request (to polycom). Alternately your
app might be configurable to
Ok,
First, thank you for your quick answer.
Now what I want to know is :
What is the TLS and SSL certificate doing in the interface is it only for
the registration of phone in the company?
Is there a version of Sipxecs that allow me to do this?
What can I use to secure the network enought to
VPN.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Simon Brûlé
Sent: Friday, March 23, 2012 10:26 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] TLS Integration (P.S) I know I posted it on the
forum and
Oops got my general meaning tho...
On Mar 23, 2012 1:27 PM, pscheep...@epo.org wrote:
Thought about that as well, only A, B and C should transfer to 2 to make
it understandablestill a cludge maybe.
Paul
Tony Graziano tgrazi...@myitdepartment.net wrote on 23-03-2012 16:19:03:
Sounds
I will have to look into this as to trying to get AudioCodes on the phone is
almost impossible. The support can be costly in the long run however I have
replacement of unit in that as well in the case of failure.
Perhaps having a Configuration Option in sipXecs can be considered?
Forgive me
I am not sure of the budget or other factors involved, however, have you
considered an inGate SIParator? I have used that device often for remote SIP
connections if VPN is not a viable option for the remote user. This is not the
normal situation though as VPN is often the safest and cheapest
As a reseller, I get excellent support on audiocodes from the Distributor as
needed. Check with your distributor.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Robert
Schroeder
Sent: Friday, March 23, 2012 10:38 AM
To:
VPN or possibly a session border controller with TLS capabilities.
On Fri, Mar 23, 2012 at 12:43 PM, Robert Schroeder
robert.schroe...@memberfirstmortgage.com wrote:
I am not sure of the budget or other factors involved, however, have you
considered an inGate SIParator? I have used that
On 3/23/2012 2:11 PM, Josh Patten wrote:
VPN or possibly a session border controller with TLS capabilities.
I have not had any success with TLS getting through firewalls or access
points that have ALG or Verizon MiFi.
--
Regards
--
Gerald Drouillard
In gate can secure with tls from outside but hand of without tls inside
On Mar 23, 2012 2:13 PM, Josh Patten jpat...@ezuce.com wrote:
VPN or possibly a session border controller with TLS capabilities.
On Fri, Mar 23, 2012 at 12:43 PM, Robert Schroeder
robert.schroe...@memberfirstmortgage.com
Thank you guys for your quick answer if we really need it we are going to
go for the VPN option since like you said its the cheapest but for now its
gone on the backburner because I just learn that the android user use
external method to do it like google voice and google talk.
So big thanks to
Hi
I try to put the auto-attendant in french and english by redirecting the
main voice menu to an other dial plan by giving him the option LOCALE=
and I also tryed with a phantom user and it doesn't work either.
I am in Québec so I really need this bilingual voice menu to work.
I also have an
Verizon mifi sux raw eggs with most vpns.
On Mar 23, 2012 2:43 PM, Gerald Drouillard gerryl...@drouillard.ca
wrote:
On 3/23/2012 2:11 PM, Josh Patten wrote:
VPN or possibly a session border controller with TLS capabilities.
I have not had any success with TLS getting through firewalls or
Yes you can vpn from an android handset and use the built in sip account or
any other compatible applications to reach the phone system. Please no, the
features you would get with what ever applications you use 4 v o I p differ
from software providers.
On Mar 23, 2012 3:14 PM, Simon Brûlé
I know its not the same thing but I am just saying that they are already
using something else for this kind of call. I was looking for patching
something that was more of a PLUS then a need so I will use my time on
something else and they will use whatever they always used. You might think
im
With the hierarchy of Polycom, changes you make on the phone, will override
the configuration file applied by sipXecs. So, making a change in a group
setting should have no affect theoretically.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On
if you honestly want flawless service.. i recommend getting an Ingate
SIParator SBC with the Remote NAT Traversal Module, SIP Trunking Module,
and enough CAL's to cover all call flows... i have found that VoIP.ms with
a SIParator provides FLAWLESS service to SipX. while having to register to
I suggested in the app. The existing polycom phones are not anything more
in the very simple browser.
Now some of the android desk and thats are a totally different story
On Mar 23, 2012 6:02 PM, Chris Rawlings cm.rawli...@gmail.com wrote:
polycom does not work with Javascript so doing
Haha. Just read my transcribed message. That's funny.
The trend will take a while, in one-two electronic generations of phones
they will have apps. Real ones. Right now we see the equivalent of the
original Motorola RAZR when they had a basic browser.
In a couple of years the android (windows or
Hmmm... In my configuration, I have two lines defined; the first with three
appearances and the second with two, then one speed dial with BLF. The speed
dial with BLF is defined in the settings for the first line. I'm not varying
from the default configuration very much, if at all.
I am
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