I've never used the Sipura phones but they probably sync with an NTP server.
My guess is that the NTP server is on the asterisk box (you can probably verify
this by checking the config of the phones and finding the option for NTP
server). It is possible that the NTP service isn't running on the
Gordon,
My guess is that you're a contractor so I can understand why you'd want to keep
yourself in high demand by steering clear of the methods that simplify
deployment and redeployment.
As an employee on the other hand, I want to make things as easy and integrated
as I can in order to
Two separate networks? Did I miss something? I feel like I'm taking crazy
pills! Two separate physical networks means twice the hassle, twice the
maintenance, twice the cost, twice the headache. Not to mention the fact that
the whole idea of VOIP is to simplify IT and focus on converging data
is given priority.
Dave
David Gibbons wrote:
Two separate networks? Did I miss something? I feel like I'm taking crazy
pills! Two separate physical networks means twice the hassle, twice the
maintenance, twice the cost, twice the headache. Not to mention the fact that
the whole idea of VOIP
You can use the Cisco phones with either the SIP of the SCCP image.
Though I do agree that the SIP image is a bit easier to setup and auto-
provision, the SCCP image is a more native (obviously) implementation.
The chan-sccp-b project has nearly every feature usable on these
phones working
Gibbons
Subject: Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME)
On Thu, Oct 23, 2008 at 10:57 PM, David Gibbons [EMAIL PROTECTED] wrote:
Dare I ask why you want to do this?
Dave
I know it seems counter intuitive but I've several examples of it
being done and for me it would
Dare I ask why you want to do this?
Dave
On Oct 23, 2008, at 10:00 PM, Stephen Reese wrote:
I was thinking about complicating my Voip setup by adding CME. I found
this example here:
http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration
and here:
wrote:
Hi Dave,
I don't view nothing in tftp server because the phone is stopped on
start
screen with displayed 'upgrading' and MAC address..I don't understand
what
happened after the reset. phone
Regards.
--
Salvatore.
- Original Message -
From: David Gibbons [EMAIL
but the reset
process is stopped !
Thanks.
--
Salvatore.
- Original Message -
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 09, 2008 4:53 PM
Subject: Re: [asterisk-users
-
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 13, 2008 3:29 PM
Subject: Re: [asterisk-users] Cisco 7906g SIP
When the 'upgrading' process fails, it means that one or more of the
required
You need to check out the chan_sccp-b mainling lists on sourceforge. There is
active development in SVN but not in tarball releases.
http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion
It is very stable.
Dave
-Original Message-
From: [EMAIL PROTECTED]
use Cisco7941 without callmanager software but only with SIP support.
Thanks.
--
Salvatore.
- Original Message -
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 09, 2008 2:30
Sasa,
You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't
have (to my knowledge) any non-callmanager SIP software. The SIP load is just a
SIP load, not a SIP load unique to generic SIP or callmanager.
Dave
-Original Message-
From: [EMAIL PROTECTED]
.
- Original Message -
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 09, 2008 2:59 PM
Subject: Re: [asterisk-users] Cisco 7906g SIP
Sasa,
Sometimes I have to do a hard reset
Did you check sip.conf to make sure that the port is correctly set to 5060?
Please show the output of Cli sip show peer peernumber and the contents of
your SEPMAC.cnf file.
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent:
We've been EXTREMELY happy with the HP 5400ZL series chassis switch. Price per
port is about 1/3 that of Cisco when it comes to POE. Price is about $100 per
port and all ports are 1Gb with POE by default -- you can't get modules that
don't have 1Gb and POE. 10Gb uplinks are available with other
Obviously we don't need 1Gb connections for VOIP :)
Phones support pass through to the desktop and VLAN tagging.
The need for 1Gb ports comes from wanting to have 1Gb at the desktop.
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson
but then you would not need the 1G uplink.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
David Gibbons
Sent: Monday, October 06, 2008 11:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PoE switch recommendations
Sean,
Try 'sip show channels' or 'sip show channel channelid' for the drill down. I
believe the codec in use will be displayed with either command.
Dave
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of sean darcy [EMAIL
PROTECTED]
Sent: Thursday,
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