Anyone else also have some input?
I just can't see why this could not be done. After all, I'm already doing it
using SIP trunks and an asterisk box. I can call someone or send a fax to the
same DID.
All I really need is to find a way of either using sipxecs to re-route the fax
to another devi
I also have access to an MC3810, is it the IOS or the hardware? Would this one
work any better?
On Tue, 11 Aug 2009 17:39:19 -0700, Max Clark wrote:
> I've been happy with c2800nm-ipvoicek9-mz.124-25a.bin Honestly after
>
> struggling with the 5350 (again for REFER support) the 2800 dropped
> r
I've been happy with c2800nm-ipvoicek9-mz.124-25a.bin Honestly after
struggling with the 5350 (again for REFER support) the 2800 dropped
right in and just worked. Haven't touched it since.
On Tue, Aug 11, 2009 at 10:53 AM, li...@grounded.net wrote:
> On Tue, 11 Aug 2009 10:20:04 -0700, Max Clark w
Assuming they have registered before sending the profile? Are the phones local
or remote?
>>> "IT Services" 08/11/09 8:19 PM >>>
Hi all:
I am using 4.01 from a CD install on a single server.
When I send a new phone profile, the phones (snom 320 or LG-6812) do not
restart automatically and henc
Hi all:
I am using 4.01 from a CD install on a single server.
When I send a new phone profile, the phones (snom 320 or LG-6812) do not
restart automatically and hence, I get a failed message in the job
status area.
Any suggestion on where to start troubleshooting?
Thanks... tommy
Notice of C
Audio codes Mediant 1000 can handle the splitting of t.38 calls from your
voice calls. They can route it to a fax server, or they have an integrated
fax server which is basically Windoze 2003 on a card. It's their own
product, so it should integrate well.
T.38 has its issues over IP, in any impl
Sn4960
-Original Message-
From: Tony Graziano
To:
To:
Sent: 8/11/2009 4:56:28 PM
Subject: Re: [sipx-users] Dealing with combined voice/faxing
4960
-Original Message-
From: "li...@grounded.net"
To:
To: sipx-users
Sent: 8/11/2009 4:54:14 PM
Subject: Re: [sipx-users] Deali
4960
-Original Message-
From: "li...@grounded.net"
To:
To: sipx-users
Sent: 8/11/2009 4:54:14 PM
Subject: Re: [sipx-users] Dealing with combined voice/faxing
What model should I be looking for?
On Tue, 11 Aug 2009 16:33:27 -0400, Tony Graziano wrote:
> Several things to bring up..
What model should I be looking for?
On Tue, 11 Aug 2009 16:33:27 -0400, Tony Graziano wrote:
> Several things to bring up...
>
> Not a "great idea" to try to separate voice/fax calls. Using separate DID's
> would be better.
> Use a Patton PRI gateway, they work tremendously well. You can take c
> Not a "great idea" to try to separate voice/fax calls. Using separate DID's
> would be better.
That's just it, right now, users are used to being able to use just the single
DID for both voice and fax calls and that to me, really is unified
communications.
Of course, the faxing portion takes u
Several things to bring up...
Not a "great idea" to try to separate voice/fax calls. Using separate DID's
would be better.
Use a Patton PRI gateway, they work tremendously well. You can take certain DID
numbers and re-route to a FXS gateway, which could in turn be used to dial a
Multi-tech digit
I forgot to mention that while some of the users will be small groups, most are
just individual users, not like an office where everyone could have access to
say a central fax server.
Mike
___
sipx-users mailing list sipx-users@list.sipfoundry.org
Lis
I'm hoping someone on this list can help clear some things up for me.
I purchased the eBook and have read everything I can find online but faxing is
still not clear to me.
We've had good results testing combined voice/fax calls to DIDs using an
asterisk flavor. However, none of the asterisk flav
My point is it doesnt do this for my polycom using the gn9350e and enhanced
blf. So this makes me wonder if it is not a plantronics setting or
configuration change. Did you check plantronics for a known issue or check with
them for the settings they suggest for e-blf
>>> Josh Patten 08/11/09 4
Correction: Enhanced BLF document from ploycom:
http://knowledgebase.polycom.com/kb/search.do?cmd=displayKC&docType=kc&externalId=EndUser-TechAlerts-Audio-SoundPointIPEnhancedBLFQT37381pdf&sliceId=pdfPage_1&dialogID=3465812&stateId=0%200%203463241
On Tue, Aug 11, 2009 at 3:00 PM, Josh Patten wrote
Let me reiterate that the EHS functionality of the plantronics device
works fine. The problem is BLF buttons in the ringing state will cause
the phone to go into the "alerting" state and the plantronics headset
will beep in the users ear. This is due to firmware version 3.1 and up
having enhanced B
I might suggest the proper settings for the polycom phone when using that
headset but i dont know what they are.
I know what they are for the GN9350e though...
http://www.myitdepartment.net/content/view/48/70/
>>> 08/11/09 3:37 PM >>>
Yes, plantronics mode on the polycoms coupled with the APP-5
Yes, plantronics mode on the polycoms coupled with the APP-5 ehs adapter
works just fine, but BLF ringing alerts cause the ehs adapter to send a
ringing state to the plantronics headset which causes the wireless headset
to emit a beeping sounds. This behavior is correct for registered lines,
Is it supported with polycom? We use the n9350e with the ehs cable and have no
issues.
-Original Message-
From: Josh Patten
To:
Sent: 8/11/2009 2:50:42 PM
Subject: [sipx-users] Plantronics APP-5 BLF woes
Problem: Whenever a BLF (busy lamp field) extension monitoring button
on a poly
Problem: Whenever a BLF (busy lamp field) extension monitoring button
on a polycom phone with a Plantronics APP-5 installed shows a
monitored extension as ringing, the APP-5 will signal ringing to the
plantronics headset device even though the phone doesn't ring.
The preferred behavior is that the
On Tue, 11 Aug 2009 10:20:04 -0700, Max Clark wrote:
> We just moved from the 5350 over to the 2800 ISR platform. The 5350
> would not support call transfers/MOH properly with sipx.
Do you think this might have been IOS related over platform? Also, which ISO do
you recommend on the 2800 if that's
We just moved from the 5350 over to the 2800 ISR platform. The 5350
would not support call transfers/MOH properly with sipx.
-Max
On Mon, Aug 10, 2009 at 9:38 PM, li...@grounded.net wrote:
> Been reading the wiki and other things and have not been able to confirm
> which cisco IOS is the best wh
Dan,
I've been using ClarkConnect since version 4.3, it works really well! I have
version 5 installed right now
and am utilizing the bandwidth management. So far, so good! It's easy to
use, manage and has a
great dashboard view!
Alan
On Tue, Aug 11, 2009 at 10:39 AM, wrote:
> Send sipx-users mai
Outgoing calls are working properly. However, for incoming calls, the call is
answered by AA, when I key in extension, the phone is dropped from the caller,
but extension still rings. Anybody has idea what the problem is?
I'm using sipX 4.0.1 ISO, SIP trunking provider is FreePBX (owned by
band
I brought this up a couple times in the past. Not only is codec support
important but also the ability to limit a certain number of concurrent
calls across a particular WAN link.
Certainly the addition of the locations concept was a step in the right
direction. This enables the administrator to
On Tue, Aug 11, 2009 at 7:36 AM, Frank J
Bruzzaniti wrote:
> If I configure a snom phone and am using it locally it works fine.
> If I move it to another location like home where I am running ADSL.
> I can connect and call an external number but there is no sound.
> I configured the RTP start and f
I have to agree with Keith about the importance of such a feature
regardless of how it's implemented. As an integrator I simply need
this tool at my disposal.
On Aug 10, 2009, at 3:51 AM, Keith Gearty wrote:
> Todd Hodgen wrote:
>
> Dale Worley wrote:
>
>> A particular problem is that in ord
If I configure a snom phone and am using it locally it works fine.
If I move it to another location like home where I am running ADSL.
I can connect and call an external number but there is no sound.
I configured the RTP start and finish range to be 3-31000 and opend
those UDP ports on the fir
On the commercial side, the Cisco ASA series can do this.
For Open Source you can look to pfsense. Vyatta will do it as well but
it is more difficult to manage.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Dan White
Sent
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