I was afraid of an unavailable NFS mount hanging the app and I also
wanted to keep all of the communication over IAX for simplicity sake. I
also hacked together my own MWI over IAX. I did write ups of how I
did both.
http://opensourcemadness.blogspot.com/2007/03/centralizing-asterisk-voic
We have several Sangoma cards that we used during a transition time in
our the replacement of our legacy voice system that we no longer need.
Each of them saw about a month of service and are in good working order.
We'd be happy to get 70% of retail for them. They are as follows:
qty 2 A104D
Do you see anything in /var/log/messages? I am having a similar problem but
I'm also getting some pci fatal error! messages. I had sangoma connect to
the box and he couldn't find any config errors so we're leaning towards a
hardware problem.
- Jeremy
-Original Message-
From: [EMAIL
, Jeremy M. [EMAIL PROTECTED] wrote:
Do you see anything in /var/log/messages? I am having a similar
problem but I'm also getting some pci fatal error! messages. I had sangoma
connect to the box and he couldn't find any config errors so we're leaning
towards a hardware problem
Are there any scripts out there that would help me stress test two boxes
that are setup back to back with 4 PRI connections? We're having
problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm
tired of testing them in a production environment. As Sangoma
provides firmware updates
We're having a lot of D channel problems with the pci-e on HP servers.
Going to PCI fixed the problem. Sangoma is aware of the problem and is
using one of our servers to work toward a solution.
-Jeremy
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I have two asterisk boxes (1.2.14) connected via SIP with Polycom 501s
registered to each. I set callerid name and num before sending the call
from one box to another but the phone registered to the receiving server
only properly shows the caller name, not the number. The number on the
phone
and CallerID
Porier, Jeremy M. wrote:
I have two asterisk boxes (1.2.14) connected via SIP with Polycom 501s
registered to each. I set callerid name and num before sending the call
from one box to another but the phone registered to the receiving server
only properly shows the caller name
While not what you are specifically requesting, making a call after a
voicemail is left is covered at
http://opensourcemadness.blogspot.com/2007/03/propagating-asterisk-mwi-a
cross.html
Using those techniques you can setup what you are describing. Rather
than calling another Asterisk server,
We are having intermittent problems where the person we call reports
static when we place an outgoing PSTN call. Only the person called
hears static, to us the conversation sounds fine. Never happens on
inbound calls. It doesn't matter what channel you call from (IAX, SIP,
or Zap). We have a
I would chip in on that bounty, but would rather see it dump to a
voicemail folder under the same IMAP account as my email.
- Jeremy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, August 28, 2006 3:31 PM
To: Asterisk
They're not the only ones :-)
Jeremy Porier
Senior Director of Information Systems and Technology
Colorado Christian University
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Wednesday, September 13, 2006 10:52 AM
To:
Looks like we will be forced to make a move on our voicemail system as
Nortel has declared Meridian Mail an end of life product. Frustrating
thing is that it would seem their only reason for it is so they can
force our hand to move to Call Pilot.
Is there any documentation and feedback out there
Brian,
While I can't say we've used this specific product, I can say that anything
we have used from RAD has been outstanding and highly reliable.
http://www.rad-direct.com/App-Ethernet-extender-copper.htm?menuId2=Applicati
onMenumenuId=Extenders2
For a season pass or two I'll come help you
We are about to deploy six Asterisk servers across the state with SIP
phones at each site registering to their local server. However, we
are centralizing voicemail at our main campus to enable the transfer of
voicemails between users regardless of site. It also simplifies our
backup procedures
boxes. We're using ssh,
you may choose to use a different method. It's an immediate MWI
notification, and seems to work well. If you're interested, let me
know, I'll shoot the scripts over to you.
On Wed, 2006-12-06 at 09:20 -0700, Porier, Jeremy M. wrote:
We are about to deploy six Asterisk
What kind of luck are people having with the Web-MeetMe control? The
condition of the page on the voip-info wiki makes me a bit nervous about
putting Web-MeetMe into a production environment. Use of MeetMe has
really taken off here since installation and I need a scheduling and
provisioning
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