I see it now, Mike has a part number for a small UPS for that phone - added
value, added sale!
Great ideas.
BTW, if you run into any Nortel BCM's, my nephew has now done
interoperability between them and sipXecs. I'm working on a document for it
now.
-Original Message-
From: sipx-users-
Yea, I'm not so sure the other phones will see DND, but at least the
console does.
These settings are found by clicking on a line that is assigned to the
phone, select Diversion, and then look for the section 'On Do Not
Distrub'.
The only 'gotcha' is that if the phone loses power the phone does n
On Wed, Oct 21, 2009 at 8:12 PM, Josh Patten wrote:
> I've never seen DND cause a BLF to light up.
Perhaps I spoke prematurely, that was an assumption on my part. It
"should" though. I'm actually more concerned that the attendant knows
that status of the phone system. This method provides that
I've never seen DND cause a BLF to light up. When I press DND on my
phone at the office, people monitoring my extension don't get the red
busy light, the BLF button acts like I'm idle. Am I doing something wrong?
If there is a way to make pressing the DND button cause a phone to
appear busy, I'
I did ask people to vote for it (in the original post earlier today) and
I have already commented on it in JIRA. I don't subscribe to the dev
list for the very same reason it took me so long to subscribe to the
user list: that whole cluttered inbox thing (among other things).
If there were a wa
On Wed, Oct 21, 2009 at 7:32 PM, Picher, Michael
wrote:
> There's yet another way to do this also with Polycom phones that I have
> used. In the Polycom settings for a phone you can re-route a call when
> DND is enabled on the phone. Thus when receptionist puts their phone
> into DND that acts m
There's yet another way to do this also with Polycom phones that I have
used. In the Polycom settings for a phone you can re-route a call when
DND is enabled on the phone. Thus when receptionist puts their phone
into DND that acts much like a night mode button.
Mike
-Original Message-
F
I've got this working.
What you need to do is setup dial plan entries to dial the other PBX's
as per these instructions:
http://sipx-wiki.calivia.com/index.php/HowTo_interconnect_two_sipX_PBXs
The trick though is to make the PBX's resolve to the external IP's of
the far end PBX.
So, sipx.domain1
Asking people to vote for the feature would help.
I went in a new office with a comdial system last week and they had a phone
setup just for activating night mode too.
Adding the feature is not as simple as editing a conf file in freeswitch.
Just because freeswitch can do something doesn't exactl
I suppose I could do something like this or put a cheap Polycom phone in
place with forwarding rules set on it for both incoming and do not
disturb, but it seems like a waste of money to buy one phone for that
purpose, not to mention expecting it to stay powered on, etc. It just
seems like a ha
On Wed, 2009-10-21 at 11:32 -0500, Josh Patten wrote:
> I realize there is an issue in Jira pertaining to this (
> http://track.sipfoundry.org/browse/XX-4821 ) but I was wondering if any
> thought had been put into implementing user invokable day/night auto
> attendant services. It really makes
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Usres Forum
X-FUDforum: 901ee31943c00dbf0827b2b0b7f984b8 <567>
Message-ID: <237.4adf3...@192.168.15.150>
Well after the last round of people requesting a
forum...I've gone and created one.
Now the
I thought this would be problematic. In the one instance we implemented it,
no issues. They had the "night mode" feature on their Comdial system before
sipXecs.
Departmentally there can be one phone per department to do this. If the
phone loses power, that would be a problem. With a Polycom phone
While this is possible, there is also a chance that the user will screw
it up or close it accidentally. Also, it does not address departments
where the last person to leave "turns out the lights". A button on the
phone would be ideal for this situation, that way they just touch it and go.
Also,
I'm downgrading them from 3.2.1 to 3.1.3RevC now.
Ken
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Wednesday, October 21, 2009 2:29 PM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: FW: [sipx-users] Best B2BUA?
Polycom firmware has a bug. I posted about that ea
Polycom firmware has a bug. I posted about that earlier today. We have an
external JIRA issue opened for that with polycom.
Did you apply the patch that is referred to in the SIPXBridge configuration
and setup page?
Regards
Ranga
On Wed, Oct 21, 2009 at 3:20 PM, Ken Fulmer <
kenful...@icstechno
We can now perform call transfers but Call Hold fails for inbound calls.
When the call is placed on hold, it cannot be resumed.
We've tried enabling MoH on the sipxbridge and disabling it on the Polycom
phones. We've also enabled MoH on the phones and disabled it on the
sipxbridge. Neither way
Thanks for your assistance. We realized we weren't pointing to port 5080.
Once we changed this from 5060 to 5080, it works.
Thanks again and I hope this helps someone else down the line!
Ken
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Wednesday, October 21, 2009 10:46 AM
To: K
With all these options I smeall a wiki update lurking...
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 Contr
We have accomplished this by allowing the hunt group to follow forwarding rules
for each ext.
Then on the main reception ext we setup a forwarding rule that they can simple
enable or disable via their user portal.
The forwarding rule just forwards right to the AA.
Works well as long as the
On Wed, Oct 21, 2009 at 1:38 PM, Mark Eissler wrote:
> I've taken a look at sipX a few times over the past couple of years to
> see if it could replace an Asterisk-based system. Here are my thoughts.
>
> When people say "Trixbox" they really mean two things: 1) the ISO; 2)
> the FreePBX GUI. The
On Tue, Oct 20, 2009 at 5:05 AM, Keith Gearty wrote:
> Picher, Michael wrote:
>
> >There's no doubt that Trixbox is good for very small installations. It
> >has a lot of functionality and can be easy to setup. For those two
> >reasons alone it can make a lot of sense for home users.
> >
> >For
I've taken a look at sipX a few times over the past couple of years to
see if it could replace an Asterisk-based system. Here are my thoughts.
When people say "Trixbox" they really mean two things: 1) the ISO; 2)
the FreePBX GUI. The drag about Asterisk is that until GUIs came along
the thing was
There is a simple workaround for this.
Forward your incoming calls to a Counterpath soft phone, set up the
Counterpath to forward to your receptionist or auto attendant.
Setup a forward rule for the Counterpath extension for when it doesn't
answer.
When you have the client open, it follows th
Setup gateway as unmanaged gateway, using DOMAIN name of the far end in the
address field.
Setup dialplan under system dialplans - define a new site-to-site dial rule
Under System, Internet Calling, turn that off.
Biggest issue is ensuring the firewall has the ports open for audio, video.
Works g
I realize there is an issue in Jira pertaining to this (
http://track.sipfoundry.org/browse/XX-4821 ) but I was wondering if any
thought had been put into implementing user invokable day/night auto
attendant services. It really makes sense for departments with variable
hours.
I don't think the
It might be a good idea to defer updating the firmware on your polycom phone
if you are using sipxbridge.
Please note the following JIRA issue.
http://track.sipfoundry.org/browse/XX-6779
Regards,
Ranga
--
M. Ranganathan
___
sipx-users mailing list si
On Wed, Oct 21, 2009 at 11:41 AM, Ken Fulmer <
kenful...@icstechnologysolutions.com> wrote:
> Anyone have a recommendation for a solid B2BUA for the sipX system? The
> built-in version seems to only work for outbound calls. We need a device
> that terminates and re-originates signaling / media in
Anyone have a recommendation for a solid B2BUA for the sipX system? The
built-in version seems to only work for outbound calls. We need a device
that terminates and re-originates signaling / media in both directions.
Thanks,
Ken
___
sipx-user
On Wed, Oct 21, 2009 at 9:02 AM, Mike J. Howarth wrote:
>
> On Wed, 2009-10-21 at 08:15 -0400, Scott wrote:
> > On Wed, 2009-10-21 at 12:43 +0100, Mike wrote:
> > > Hi,
> > >
> > > Relatively new to SipX, and having a few problems with inter-pbx
> > > connections.
> > >
> > > What I want to achie
Anyone know if the 2nd line on the ATA 186 is usable in sipX?
In the Cisco CM, you can drop the first two digits and add 01 to the end and
it registers the 2nd line as a separate phone. However, I'm not getting the
same result on the sipX server.
Any ideas?
Thanks,
Ken
__
On Wed, 2009-10-21 at 08:15 -0400, Scott wrote:
> On Wed, 2009-10-21 at 12:43 +0100, Mike wrote:
> > Hi,
> >
> > Relatively new to SipX, and having a few problems with inter-pbx
> > connections.
> >
> > What I want to achieve is :
> >
> > SipX-A through SIP Trunk to SipX-B using SBC's - becaus
-Original Message-
On Wed, 2009-10-21 at 12:43 +0100, Mike wrote:
> Hi,
>
> Relatively new to SipX, and having a few problems with inter-pbx
> connections.
>
> What I want to achieve is :
>
> SipX-A through SIP Trunk to SipX-B using SBC's - because these will
> ultimately be behind f
The Polycom KWS 300 and KWS 6000 are both known to work well with sipXecs.
("Registered Product" status for Nortel SCS -
http://www.nortel.com/prd/dpp/product/scs.html#P)
-Paul
paul.moss...@nortel.com
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:
On Wed, 2009-10-21 at 08:15 -0400, Scott Lawrence wrote:
> On Wed, 2009-10-21 at 12:43 +0100, Mike J. Howarth wrote:
> > Hi,
> >
> > Relatively new to SipX, and having a few problems with inter-pbx
> > connections.
> >
> > What I want to achieve is :
> >
> > SipX-A through SIP Trunk to SipX-B u
On Wed, 2009-10-21 at 12:43 +0100, Mike J. Howarth wrote:
> Hi,
>
> Relatively new to SipX, and having a few problems with inter-pbx
> connections.
>
> What I want to achieve is :
>
> SipX-A through SIP Trunk to SipX-B using SBC’s – because these will
> ultimately be behind firewalls.
>
> I’ve
Hi,
Relatively new to SipX, and having a few problems with inter-pbx
connections.
What I want to achieve is :
SipX-A through SIP Trunk to SipX-B using SBC's - because these will
ultimately be behind firewalls.
I've gone through steps to define what I think needs defining and am now
seeing packet
On Tue, 2009-10-20 at 17:27 -0700, Bryan Simmons wrote:
> Does anyone know where the raw CDR records are kept on the SIPX?
See
http://sipxecs.sipfoundry.org/rep/sipXecs/branches/4.0/sipXproxy/doc/cdr/cdr_user_guide.pdf
___
sipx-users mailing list
Searching through audiocodes manual for "ringback" helped me some time ago.
Nikolay.
_
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of heros
Sent: Wednesday, October 21, 2009 3:09 PM
To: sipx-users@list.sipfoundry.org
Subject:
Again on audiocodes! I have a MP-124 FXS, 24 port. Telephones are registered
on sipX they can both place and receive calls BUT..no call progress tones
are heard by the user. I suspect this is connected to CPT tones. Problem is
that I don't know How to generate CPT tones for audiocodes.
Heros
I tried Nokia E65 and it is good with sipX. I suppose all E6X have similar
performances.
About DECT I would suggest Kirk server v500 as a good DECT system for big
deployements, but I tried it with asterisk not with sipX.
Heros
-Messaggio originale-
Da: Keith Gearty [mailto:ke...@glens
On Wed, Oct 21, 2009 at 6:02 AM, Picher, Michael
wrote:
> Also, make sure they are sending you calls on port 5080udp They like
> to send on 5060.
>
>
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
> *Sent:* T
Also, make sure they are sending you calls on port 5080udp They
like to send on 5060.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Tuesday, October 20, 2009 2:15 PM
To: jermaine pinder
Cc: sipx-users@list.sipf
On Wed, Oct 21, 2009 at 4:28 AM, Keith Gearty wrote:
> Josh Patten wrote:
>
> >I am sure this question has been asked before, but I'll ask again in
> >case new information is available:
> >
> >Has anyone had good experience with any particular DECT/Wi-FI phones and
> >sipX? Does sipXconfig have a
Josh Patten wrote:
>I am sure this question has been asked before, but I'll ask again in
>case new information is available:
>
>Has anyone had good experience with any particular DECT/Wi-FI phones and
>sipX? Does sipXconfig have any management capabilities for any of these
>phones? (my guess is
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