On Tue, 2009-07-21 at 17:12 -0400, ingo...@netvision.an wrote:
> What steps should one take to have BLF on Soundpoint phones? Preferred
> result is that a user can see his/her contact status at all times without
> even attempting to call him/her.
This Wiki page describes how to set up the BLF list
is there any need of sip trunking service providers leased line to test sip
trunking between two local IPPBX's? if yes should i get any test acount
with any service provider?
On Tue, Jul 21, 2009 at 7:28 AM, Tony Graziano wrote:
> Yes. If the two LAN are interconnected without a firewall or vi
Tony,
Thanks for the quick reply! The 5800 port was a typo (sorry) it is indeed
5080 (checked and double checked firewall and sipx config). The ITSP is a
company called Binary Telecom (www.binarytelecom.com). They are a local
company and I have direct contact with the owner (thought that might b
A little more information would be helpful. What carrier are you using for
siptrunking? The issues you are mentioning suggest a trunking issue rather than
firewall. A lot of ITSP's do not support every feature.
You mentioned port 5800, normally that would be 5080. Is 5800 the signalling
port you
Hello,
I am attempting to use sipx as a solution for a small business telephone
system. I have 10 Polycom IP330's and one Polycom IP650.
I would like to have the following configuration:
- Firewall (either m0n0wall or PFSense) on a DSL circuit
- SIP Trunk using sipxbridge
- Support t
What steps should one take to have BLF on Soundpoint phones? Preferred
result is that a user can see his/her contact status at all times without
even attempting to call him/her.
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On Tue, 2009-07-21 at 12:44 -0500, Goran Donev wrote:
> As we all have probably heard that Avaya is buying Nortel’s Enterprise
> division.
>
> Does that mean Avaya developers will be joining the foray of SIPX
> development?
http://list.sipfoundry.org/archive/sipx-users/msg15960.html
__
As we all have probably heard that Avaya is buying Nortel's Enterprise
division.
Does that mean Avaya developers will be joining the foray of SIPX
development?
Thanks.
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I've gotten a few off-list requests for information about yesterdays
announcements regarding the sale of the Nortel Enterprise business.
For anyone not following them - Nortel[1] and Avaya[2] announced
yesterday that a bid from Avaya has been accepted as the so-called
'stalking horse' offer to pur
Hello All,
We currently have a environment with 3 x Sipxecs 4.0, 1 server being main
server holding AA, dial plans, and hold 1 departmental queues. The other two
are holding 3 departmental queues each.
We were having issues upgrading to Sipxecs 4.01 via yum process, so we made a
plan to upg
OK, no problem, will sent to you soon. Thanks in advance.
-Original Message-
From: Robert Joly [mailto:rj...@nortel.com]
Sent: Tuesday, July 21, 2009 10:44 PM
To: jun,wen; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Remote worker cannot call to second intranet
subnets
Ok, I
Ok, I took that as far as I could without a snapshot. Could you turn
the logging level to DEBUG for sipXproxy, reproduce the call and send me
your system snapshot?
> -Original Message-
> From: jun,wen [mailto:jun@msn.com]
> Sent: Tuesday, July 21, 2009 10:41 AM
> To: Joly, Robert (
Yes, it is still one-way speech path, from remote worker to sip client
inside intranet. The router well terminates VPN tunnel for these two subnets
since sip phone inside them can make call to each other. Both of them are
disabled SIP ALG.
-Original Message-
From: Robert Joly [mailto:rj...
Ok, we are making progress. Still you get one-way speech path? What
about the routers terminating the VPN tunnels for 192.168.8.0/24 and
192.168.10.0/24? Would they also have a SIP ALG enabled?
> -Original Message-
> From: jun,wen [mailto:jun@msn.com]
> Sent: Tuesday, July 21, 20
I made my eyebeam as you pointed. Now it gives me "x-sipX-privcontact" in
the Registration.
-Original Message-
From: Robert Joly [mailto:rj...@nortel.com]
Sent: Tuesday, July 21, 2009 9:16 PM
To: jun,wen; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Remote worker cannot call t
We got an SCS500 to finally dial out using an MP-114 FXO but it 8-10
seconds after you dial a number before you hear the ringing. The phone
being used is a Polycom IP550.Does anyone know why or what could be
causing this delay?
Any ideas or suggestions will be greatly appreciated J
Thanks
After making your changes make sure that you restart eyebeam to force a
re-registration. If your eyebeam still shows up with a x-sipX-nonat in
the registration page of sipXconfig then something is still trying to do
NAT compensation (and doing a poor job at it). For eyebeam, please make
sure that
Just add a secondary ip address to the current ip address on the Exchange server. Go to edit the ip address and then click advanced. There is a box to add a secondary ip address. Exchange 2007 will immediately start listening on all available ip's (unless you specifically
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