Eric Germann wrote:
How do you handle transfering vmail from one user to another when they're on
separate servers?
I'd have a look at:
http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage
Having voicemail stored in a database solves all kinds of potential
locking problems. I
Hi!
While a connection in progress on one extension, I would like to go to
any other phone, dial some extension number, in order to ether pick up
the call or join in an automatic conference. In other words, make it
work like the old ma bell phone (when I want it to :-) )
Is this
On 28 Apr 2007, at 22:22, Laurent CARON wrote:
Hi,
I'm wondering what the best option to obtain a high availability
asterisk server is.
Here's what we do when consulting in this area:
First decide what the maximum acceptable downtime is, and
what the costs to the business of that downtime
HI All;
I want to use Asterisk for just Voicemail Server and I need a Dynamic creation
of Mailboxes.
My users 's Mailboxes are same as Extensions but I donot want to add
mailboxes in
Voicemail.conf
Is there any way to create mailbox from Asterisk dial-plan ?
Appreciate any suggestions
On 28 Apr 2007, at 09:05, Matt Gardner wrote:
Ok this is my first post and I will try to keep it short.
I have searched everywhere and haven't found an answer to my question
I have two Trixbox servers that are connected over the Internet via
an IAX2 connection. We are experiencing very
Tim Panton wrote:
Here's what we do when consulting in this area:
First decide what the maximum acceptable downtime is, and
what the costs to the business of that downtime would be.
Use that as the starting point for the HA design.
Discuss with the telco what they can do.
They can
On 27 Apr 2007, at 19:59, Mike wrote:
Michael and all those who replied,
This Linksys WBP54G does seems to be what I need, but it also
seems very much made for Linksys phones. Isn't there some sort of
equivalent thing that comes with it's own power supply (at the cost
of needing
On 26 Apr 2007, at 21:07, Anthony Rodgers wrote:
It will stall asterisk - ask me how I know.. :-)
CP
Ouch!
NFS is great in a well regulated stable LAN with low latency and a
high read/write
ratio. Take away any of those factors and it rapidly turns into a
nightmare.
If I were
Pardon my ignorance, but:
On Sun, Apr 29, 2007 at 11:36:05AM +0100, Tim Panton wrote:
On 28 Apr 2007, at 22:22, Laurent CARON wrote:
Since it is possible with BRI cards, i'm wondering if it could be done
with PRI.
No, BRI has a 'bus' topology, PRI is point-to-point.
Taking a BRI from
On Sun, Apr 29, 2007 at 11:55:43AM +0100, Tim Panton wrote:
TP Theoretically Skype _can_ produce better audio quality than asterisk
TP as it supports a wideband codec
Asterisk 1.4 support G.722 transit, and trunk support G.722 transcoding.
--
JID: [EMAIL PROTECTED]
ICQ: 58417635 (please,
Hi List;
What is the difference between TDM11B and TDM04B? Why
the price of TDM11B cheaper than the price of TDM04B?
Regards
BIlal
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Sorry, I sent the following reply with the wrong from address and so
it did not pass the lists spam filter. So here is the message again:
Hi James,
Thank you very much for your help!
You were right, the codec is not compiled into my asterisk version. I'm
using debian, too and a show translation
On Sun, 29 Apr 2007, bilal ghayyad wrote:
Hi List;
What is the difference between TDM11B and TDM04B? Why
the price of TDM11B cheaper than the price of TDM04B?
The 11B has One incoming and One outgoing port.
The 04B has Four four incoming ports.
You need to know the difference between FXO
Hi List;
Can someone advise me if Polycom support H323 that
work fine with Asterisk? And wether this H323 Polcyom
devices more costly than SIP Polycom.
Also, I am not able to know if new Polycom come with
PoE adaptor so no need for PoE Switch (can use normal
switch that does not support PoE)? Do
I've heard of a device that acts as a failover for a PRI line so you
can plug a PRI into two different devices and have the PRI failover if
one device fails. Unfortunately nothing like this is commercially
available today.
Sounds like the ISDNguard:
Hi Eric -
How do you handle transfering vmail from one user to another when they're on
separate servers?
I'm using the single vmail server, mounted NFS partition for this right now.
I'd love to be able to have them standalone so they're survivable when the
WAN collapses, but I haven't figured
Hi Bilal -
What is the difference between TDM11B and TDM04B? Why
the price of TDM11B cheaper than the price of TDM04B?
The TDM11B has one FXS port and one FXO port. The TDM04B has four FXO
ports. More FXO modules on the TDM04B means a higher price.
- Noah
Hi Noah,
Russell from Digium is working on a piece of code for monitoring
'states' or 'status' across multiple servers (mainly to monitor do I
have a voicemail for collection on another server).
Do a google on the -dev list and you'll find some posts to reply to.
Regards,
Dean Collins
Hi,
Is it possible to have MOH in early audio, while waiting for someone to pick
up a Dial() call?
(When using zap channels, I have early audio working with playback)
Håkon Nessjøen
Loopback Systems AS
___
--Bandwidth and Colocation provided by
Who resells these products in the USA or at least ships here?
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Sune Kloppenborg Jeppesen
Sent: Saturday, April 28, 2007 7:19
The m switch should play music just as r will generate a ring.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Håkon Nessjøen
Sent: Sunday, April 29, 2007 12:28 PM
To: asterisk-users@lists.digium.com
Russell from Digium is working on a piece of code for monitoring
'states' or 'status' across multiple servers (mainly to monitor do I
have a voicemail for collection on another server).
Do a google on the -dev list and you'll find some posts to reply to.
Hey Thanks Dean! That sounds like a
Hi. I am having a problem with asterisk 1.4 and flashing a zap fxo
line on a tdm 400p. in 1.2 I was able to type *0 after flashing the
hook on my extension if it were talking to a zap fxo line, the fxo
would flash and I could pick up call waiting or whatever -- now in 1.4
this does not work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Totaro
Sent: Sunday, April 29, 2007 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Poor man's High Availability solution
Who resells
No probs there is a recording of a discussion on the 20th of April
located here
http://www.talkshoe.com/talkshoe/web/talkCast.jsp?masterId=22622talkCas
tId=22622
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
-Original Message-
From: [EMAIL
On Sunday 29 April 2007 16:25, Noah Miller wrote:
I've heard of a device that acts as a failover for a PRI line so you
can plug a PRI into two different devices and have the PRI failover if
one device fails. Unfortunately nothing like this is commercially
available today.
Sounds
i have a customer that needs to plug the phones into the pc's
using the pass-through rj45 available on most sip phones
the question they are asking me is how to keep the data network
separate from / secure from the voip network
i understand they can set up vlans but i am hazy on a few details
If you are using a cisco switch (2950, 3560, CE500, 4000, 6500, or 3750)
then you will be able to setup the phone and have the computer daisy
chained to it.
I have a similar setup on mine. Here's how I configure my switch ports
in order to achieve the desired effect:
switchport access vlan 5
On 4/28/07, Noah Miller [EMAIL PROTECTED] wrote:
Nope. You can use a device like the Redfone fonebridge to convert the
PRI to TDMoE. Possible Downside: I've read some reports that say the
TDMoE module in asterisk is not so stable.
I highly suggest you don't use TDMoE. Especially not if you
I could be wrong, but I believe that the newer Polycom's only support
SIP. The 550 is PoE, but you don't have to power it with PoE. I have a
430 that uses an A/C adapter and regular Ethernet. I have a 501 that
comes with a proprietary Ethernet cable that has an adapter built into
the cable to
The 501 is more weird then that. The cat5 cable with the built in power
injector is cool but to use it with a PoE (802.3af) switch you need a
special cable (the pairs are just different you can probably look it up
and make your own).
I believe the polycoms have MGCP and SIP support, no H323.
On Sun, 2007-04-29 at 06:00 -0700, bilal ghayyad wrote:
Hi List;
Can someone advise me if Polycom support H323 that
work fine with Asterisk? And wether this H323 Polcyom
devices more costly than SIP Polycom.
Also, I am not able to know if new Polycom come with
PoE adaptor so no need for
All,
I have a Polycom 650 phone, when turned on displays Checking
application.
Can any give me some information as to what is wrong? I have copied the
CFG files from a 601 phone to work with this 650.
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--Bandwidth and Colocation provided
What version of the SIP firmware is on you boot server? You need to check
with your reseller, because the 650 needs at least 2.1.0.
On 4/29/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
All,
I have a Polycom 650 phone, when turned on displays Checking
application.
Can any give me
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