Yes the config files would be great. Thank you very much
Thank You,
Chris Rawlings
Network Specialist
KeystoneSoftware Solutions
610.685.2111 ext 117
Follow Keystone Software on...
-Original Message-
From: Jay Coatline [mailto:jcoatl...@gmail.com]
Sent: Tuesday, September 01, 2009 7:1
Hi Chris,
I wouldn't give up on the Ciscos. We have a number of 79x5 phones
(same firmware as 79x1 I believe, and also use the CiscoPlus config)
which work very well as remote phones behind NAT.
When I am next at the site, I'll get the config and can share it with
you, if that would help.
J
On
Hey guys have a weird issue. We're running 4.0.1 and I have added 2
users extensions 221 and 222 which are Spectralink 8030 handsets. When I
try to call a polycom desk phone (Ext 246) I get "Rejected Request
timeout" on the handset. However if I dial 222 or 221 from the desk
phone (246) the handset
Oops! Sorry that was not an appropriate response. I see Matt was trying to
help.
I ought to read more carefully.
Ranga.
On Tue, Sep 1, 2009 at 1:33 PM, M. Ranganathan wrote:
> Screen shots are not very helpful. Please look at the troubleshooting
> section in that wiki page. I would need snapsh
I sent the screenshots because he is having issues with his vitelity setup and
as far as I know, there are no outstanding bugs with viteltiy in 4.0.1...so the
screenshots are mearly so he can make sure he setup the account in sipx
properly.
We have tested Vitelity with fresh 4.0.1 and with the
Screen shots are not very helpful. Please look at the troubleshooting
section in that wiki page. I would need snapshots to diagnose problems and
yes you still need the patch if you are not up to 4.02 in your ISO.
Regards
Ranga
On Tue, Sep 1, 2009 at 1:25 PM, Matt White wrote:
> I just sent you
I just sent you some screenshots offlist for a viteleity account we have at a
customer site.
-M
>>> Tim Booth 09/01/09 11:20 AM >>>
I'm struggling to get vitelity and varphonex sip trunks to register. If
they do register would it show under Diagnostics> registrations?
Are there a forums sites
Mike,
You can use the SOAP API to do this. You'll get little in the way of options,
you can create users,phone and groups, but there is no way yet to say set the
forwarding options for a user through the SOAP API.
http://sipx-wiki.calivia.com/index.php/SipX_ConfigServer_SOAP_API
Kyle
--
Ky
Hi All,
I have a problem that is *like* XX-5690
sipXconfig (4.0.1-015823 2009-06-19T00:09:44 oem-4.0-centos5)
Where I create a conference for a user 1000 with a prefix of 6 so the
conference number is 61000. Then I create an autoattendant with a prompt
asking them to enter the conference
On Tue, Sep 1, 2009 at 11:19 AM, Tim Booth wrote:
> I'm struggling to get vitelity and varphonex sip trunks to register. If
> they do register would it show under Diagnostics> registrations?
>
No it would not show there for 4.01. The settings are here :
http://sipx-wiki.calivia.com/index.php/SI
I'm struggling to get vitelity and varphonex sip trunks to register. If
they do register would it show under Diagnostics> registrations?
Are there a forums sites or just this list?
Regards,
Tim Booth
VisionCom
MaineVoIP Systems
(207) 321-2789 ext 115
tbo...@visioncom.us
---
On Tue, 2009-09-01 at 09:21 -0400, Gerald Drouillard wrote:
> When calling a Paging group can it override the user's DND setting on
> the phone?
That is going to depend on the phone. I don't know how the popular
phone makes handle that conflict, but it shouldn't be hard to test.
Dale
I'm having a difficult time figuring out how to setup a sip trunk with
Vitelity even with the template. Is anyone willing to send me a
screenshot of their setup with your info xxx out that I can use as a guide?
Regards,
Tim Booth
VisionCom
MaineVoIP Systems
(207) 321-2789 ext 115
tbo...@visio
Valcom for both. They make a timer (I don't recall the model number)
that can ring at certain intervals. Combine that will a VOLCOM SIP
paging adapter and you should have a pretty solid solution. I've used
the timers in warehouse environments for shift/lunch bells but have
not used the combo. C
1-2-3
-Original Message-
From: Tim Booth
To:
Sent: 9/1/2009 9:31:02 AM
Subject: [sipx-users] test
list test
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Valcom makes a pretty good sip capable overhead pagin system that should work
with sipx. I don't think paging or anything else can override a DND on a phone.
As far as the cron script, it would have to be a sip client. Maybe a softphone
on a pc could do this? In order to make a call on sipx you
We have sipx systems installed in a school with over 60 extensions. We
would like to upgrade the bell and paging systems now.
When calling a Paging group can it override the user's DND setting on
the phone? Or should I run in ceiling speakers to all the rooms and
have an extension to the devi
Try moving to polycom 3.1.3 (I use split & most current of 3.1.3) and
bootrom 4.1.3..
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of James
Litwin
Sent: Monday, August 31, 2009 4:14 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-us
Yes! Could be quite entertaining.
Unfortunately it was the only way to get changes from your phone back up
to the PBX. Not that I am endorsing such a configuration... I'm just
pointing out what works(ed).
Mike
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Hi all,
If I'm not mistaken, sipX at the moment (4.0.1) does not support TLS. And I
have not find it in the roadmap.
But can somebody please shed a light if it is possible to use SIP/UDP or
SIP/TCP for signaling and SRTP (if a phone can do it) for voice?
And can the snom 360/370 phone be con
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