http://www.voipsupply.com
or call at
1-800-398-VOIP
they can rush deliver if you need it.
Original Message
Subject: [Asterisk-Users] Source for Sangoma or Digum 2+ port T1 Card
near NH??
From: Tom Rymes [EMAIL PROTECTED]
Date: Fri, July 01, 2005 8:31 am
To: Asterisk Users
token authentications for secure management of the remote
locations.
Cheers,
Max W. Blackmer, Jr.
Consultant, Knowledge Power IT
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to bring the other
servers up to date.
You might also consider DNS round robin to distribute initial
connections to MySQL. A better solution is to use a load balancer that
tracks to see if the servers are alive and balance against the load on
each server.
Cheers
Max W. Blackmer, Jr
If you have licenses or experimenting with g729 you might want to look
at http://www.readytechnology.co.uk/open/g729/
it requires Intel C/C++ compiler and IPP libraries.
Original Message
Subject: Re: [Asterisk-Users] Digium G729 licensing - is it worth the
trouble?
From:
[EMAIL PROTECTED] 1.1 Released and can be downloaded from Sourceforge.
http://sourceforge.net/project/showfiles.php?group_id=123387package_id=135368
Cheers,
Max W. Blackmer
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Hello,
APC has a nice selector tool on their website.
http://www.apc.com/tools/ups_selector/index.cfm
It asks several questions to recommend an APC solution. it even gives
you a percentage of the capacity of the ups systems capacity.
Original Message
From: Wilson Pickett
into other CRM Applications.
What are your thoughts on this?
Max W. Blackmer Jr.
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Original Message
Subject: Re: [Asterisk-Users] G729 codec
From: todd [EMAIL PROTECTED]
Date: Wed, May 25, 2005 11:59 am
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Steve
Trying to understand the floating point vs fixed,
802.3af 10/100 ports
List price is $499.99 USD. I can get them for $420 USD.
DES-1526 Web-Smart 24-port PoE 10/100 + 2 Combo Gigabit Copper/SFP ports
Switch.
List price is $999. I can get them for $850.
Cheers,
Max W. Blackmer, Jr.
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it works here in Chicago. you might want to check with your provider
their dns may be out.. that happened with comcast about 3 weeks ago.
Original Message
Subject: [Asterisk-Users] Voicepulse down?
From: Trevor Harrison [EMAIL PROTECTED]
Date: Wed, May 11, 2005 8:57 am
To:
Satellite delays are always bad. It is more a delay because of the time
it takes a signal to travel to the satellite and back to a receiving
station. You might want to check into ground station to station
microwave communications stations. The best is to have a tap to a phone
company that may
.prod.mesa1.secureserver.net with SMTP;
11 May 2005 18:40:07 -
Received: (qmail 952 invoked by uid 99); 11 May 2005 18:40:07 -
Message-ID: [EMAIL PROTECTED]
Date: Wed, 11 May 2005 11:40:07 -0700
From: Max W Blackmer Jr [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Satellite Providers
To: Asterisk
Take a look at the Polycom 360 if you only nee 12 lines. otherwise look
at the Snom 220 with a sidecar (up to a total of 3 side cars may be
added for a total of 65 lines in the extreme need.)
Max W . Blackmer, Jr.
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Here is an excellent document explaining the differences between FXO and
FXS.
http://www.google.com/url?sa=Ustart=5q=http://www.patton.com/technotes/fxs_fxo.pdfe=7385
Also you can look at Digium's site for their description, which
describes it from a stand point of Asterisk as the PBX.
Linksys has a low end router with an 8 port switch that does QoS model
BEFSR81. It can be gotten for under $100 USD. For more information
http://www.linksys.com/products/product.asp?prid=604scid=29
Max W. Blackmer, Jr.
Original Message
Subject: [Asterisk-Users] 4 - 8 port w
Don't forget Dundi is such a system that is already integrated into
Asterisk.
http://www.dundi.info/
Original Message
Subject: [Asterisk-Users] BIND VoIP anyone?
From: Andres Paglayan [EMAIL PROTECTED]
Date: Thu, April 28, 2005 11:39 am
To: Asterisk Users Mailing List -
look here
http://www.polycom.com/resource_center/1,1454,pw-6812-9192,00.html
Original Message
Subject: RE: [Asterisk-Users] Polycom IP4000 Conference Phone
From: Wiley Siler [EMAIL PROTECTED]
Date: Tue, April 26, 2005 8:39 am
To: Asterisk Users Mailing List -
Now for my present delima. - Actually this one's been racking my brain since
about March.
I need to find a Cisco Reseller.
very good and will pre configure the Cisco phones for SIP.
http://www.voipsupply.com/home.php
--Snip--
Last month, I purchased a Cisco-7905G IP Phone from a vendor
spending over $A10,000 in the process. The cards are more expensive than
the server they're going into (Dell poweredge 750's). When a GPL'd hardware
It is obvious that you have never experienced high end servers. We have
had a single server cost as much as $20,000 and that is nothing but
high
There's no transcoding going on. It's ulaw on IAX with Sixtel and ulaw
on SIP to the phone. I considered that as a possibility originally, and
even tried using GSM with Sixtel to force it to do transcoding, but had
the exact same problem.
The Asterisk box is a 2.4ghz P4 with 512MB RAM,
I don't see any way to tell the Polycom to ignore QoS. It's mainly
routers and switches that pay attention to QoS, the phone would just set
QoS on its outgoing packets. Anyway, here's what's in the QoS section-
it all seems to be related to sending packets:
It is not in the transport if
with the capability of emailing or
generate on demand for web download for clients.
Any other Ideas anyone might need in addition to trabas features?
Thanks,
Max W. Blackmer, Jr.
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We recently configure an asterisk server to use with an VoIP provider
to make calls to a PSTN. We use (voipjet, nufone, diamond)
We feel that we haven't got the quality that we hope. Sometimes our
calls gets mute, or we feel communication cuts on our phone calls.
We have got an QOS
Snip
During tests with a IAX2/PSTN gateway I've been getting strange results for
processor idle time and load. I (re)search(ed) this issue for a while, but I
didn't get any good explainations. Can somebody help me?
Yes, Speex is pretty cpu intensive compared to other
You Might want to look at Trabas (
http://www.trabas.com/opensource/index.html ) it is by far the most
complete billing system.
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To
:) trabas and asterisk have big misunderstanding the don't thing to work
like it should be :)
Just needs some programming to translate asterisks logs and import them
into the database tables.
:) that is the good thing about opensource
Max
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It's going from 7.95 ( 8 bucks ) a month to 11 bucks, an increase of
72%. That's hardly what I'd call doubling ( unless you're using that
new math I've heard so much about ).
h, actually it is only a 28% increase. you want to see outrageous
you should see my gas bill.
Hello,
Yes, there is high availability, Clustering and Load Balancing. Each
one has its own advantage and disadvantages.
First option is one that you mention High Availability. This option you
have a second machine watching heartbeats from the primary machine.
when the heartbeats stop the
Just found a 12 port single card with opensource drivers
12 user configurable FX0/FXS analogue ports for $1,680 at asterisk mall
( http://www.asteriskmall.com ).
I am not sure how well this card works with asterisk. Has anyone used
these cards?
Voip supply has a few 24 port gateways that are
Voip supply has a few 24 port gateways that are FXS based. The biggest
one for FXO is 10 ports. They are not cheap the both cost about $2000
USD. a Channel bank with a T1 card will cost you about the same at
least with a FXS ports.
FXO costs more usually because that is typically the Office
Thank you John,
Max Blackmer
I would like to create an Intercom extension that will dial a group of
extensions which are connected to SIP phones. The SIP phones are setup
to auto answer a particular extension assigned to one of the lines in
the phone. All phones must answer and broadcast
at
the same time.
Has anyone done this? Or should I install an overhead speaker system
using the oss/alsa console as a broadcast. Can the local port be set to
auto answer calls?
Thank you,
Max W. Blackmer, Jr.
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do you have libtiff source files installed?
Original Message
Subject: [Asterisk-Users] Problem Compiling Spandsp
From: Juanjo Portela [EMAIL PROTECTED]
Date: Mon, March 14, 2005 6:44 pm
To: Lista Asterisk asterisk-users@lists.digium.com
Sirs,
I can't compile the source
Ronlald,
Did you use any options with the diff command?
The usual option for producing patch files is -u the unified format.
Example:
diff -u originalfile newfile patchfile
Hope this helps
Max
It bothered me, so I changed it to my need, how can I contribute this
changes back to the
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