WARNING: This has been cross-posted in the Asterisk-BIZ mailing list.
Hello, we are looking for a unix programmer who can write the
following type of application that works with asterisk.
1. Operator Console.
When a call comes in on a did, the did should be looked up in a
database(against a
Does anybody have an example follow-me configuration web page code
written in either php or perl that can write out the follow-me config
into the asterisk files?
I'd like to setup something on our office voip server that I can
change as needed via a web page rather than writing the script by
Are there any compactPCI boards that work with Asterisk?
Brent
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We like it alot. It makes rapid deployment of asterisk boxes a breeze.
Brent
On 8/17/05, Sharadindu Mohanty [EMAIL PROTECTED] wrote:
Hey Guys,
Wanted a Suggestion..Howz this Xorcom Asterisk?I am using it and till now
its fine as currently it is in testing stage with 3-4 users.
Any
How does one block registration of sip phones using asterisk if that
sip phone is on a subnet other than the one allowed?
Thanks,
Brent
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Are there any compact pci boards which asterisk supports?
Thanks,
Brent
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be unix console
based(perl-ncurses?) or Windows based.
Anybody have a line on a good asterisk programmer that could do
something like this?
Brent Clements
IO Networks
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Can you explain your thought pattern further?
On 8/4/05, Mark Phillips [EMAIL PROTECTED] wrote:
Surely This is an ethernet issue rather than a * one?
You are still left with the problem of the protocol on the trunk too.
brent clements wrote:
I have the following upcoming install
On the actual switchboard side of this, how would asterisk handle
this...going to a central person/extension as well as notifying that
person who is calling and what line they are calling on.
I have a customer right now who is hesitant to move to asterisk
because of the switchbaord situation.
Please don't hijack this thread.
On 8/4/05, Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote:
Senad,
I don't want to take this conversation much further and send a laundry
list of issues we faced with Switchware to this forum, and the myriads
of bugs we have wrestled with in the past 18
Does anyone have experience with melding Asterisk with an older Executone PBX?
I have a client whose existing voicemail server(repartee) has become
bonkers and we need to stick a VM system in there asap. I thought
asterisk would be a good thing to use.
Does anyone have experience with the older
Please stop hijacking this thread. Your little spat is not the point
of this thread.
Thanks
On 8/4/05, Senad J [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Senad,
I don't want to take this conversation much further and send a laundry
list of issues we faced with Switchware to this
I have the following upcoming install and I'm trying to do it without
having to resort to Digium t1 cards.
I have a Cisco IAD2431 being installed by our Carrier. That Carrier
will be providing 2 IP Trunks via ethernet handoff into the Cisco
IAD2431. The CIsco IAD2431 has Two T1 ports installed
it with Executone but have with others. The biggest problem is
knowing that you have a VM waiting for you. There's know way to light
the light on the PBX system.
How many nodes on the old system? If its not to many why not replace it
with * entirely? Bring them out of the dark ages.
Mark
brent
Yeah we have been planning this deployment for 4 months, we are up to
300 phones as the trial. 1k phones is a bit overwhelming.
B
On 8/4/05, Mark Phillips [EMAIL PROTECTED] wrote:
Wow!! 1000 nodes. Good luck
brent clements wrote:
Well this is a temporary fix. We are gearing up a load
For those of you who have been working with asterisk for a while and
who have experience with SIPxchange, why have you chosen Asterisk over
the latter?
What are some significant differences between the two that those of
you familiar with both have discovered?
Brent
market but don't have the money or resources to
develop or purchase a commercial product.
We paid somebody to do ours though, if there was an opensource version
when we were getting into this, we'd probably use it and also
contribute.
That's just my two cents.
Brent Clements
IO Networks
On 8/3
Hello, long time lurker, first time writer
We have the following set up
ITSP
|
|
Internet
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Cisco 2600
|
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SwitchAsterisk Server running 1.0.9(has public ip)
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Cisco 515e Pix Firewall running Pix OS 5.3(run's a class c 1-to-1 nat and pat)
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Grandstream GXP-2000(run latest fw from
What would be a the minimum hardware requirements
for a small asterisk pbx that would only have 2 pots lines coming in(2 fxo
ports)but with 4 extensions(4 fxs ports)
And enough space to hold up to a month of vmail for
those 4 extensions/users?
Heck what arethe typical hardware
requirements
Hi, I'm looking for a multi-line sip/ip phone that
can answer multiple incoming paths. IE a secretary sits at a front desk and
cananswer multiple incoming lines/DID's.
Is there something like this?
Thanks,
Brent
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I think I have idea what IAX and SIP termination
means, but can someone explain this to me?
For instance, how and why would I use someone like
iax.cc?
Thanks,
Brent
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to a proxy device via either protocol, the proxy is said to
terminate those protocols.
Brent Clements wrote:
I think I have idea what IAX and SIP termination means, but can
someone explain this to me?
For instance, how and why would I use someone like iax.cc?
Thanks,
Brent
Ok, so I'm setting up my small office.
I have my asterisk machine setup and I have 3 sip
phones connected as my stations and a 4 port FXO card ready to go(planning on
only using 2 lines currently).
What should I now order from my telco(sbc in this
case)
Everytime I call, they want to
Well...the office doesn't currently have broadband,
so vonage would be out. We don't plan on getting broadband at that office either
so...PSTN is my only access option. I'm was just curious if there was a specific
type of cheap line I should order.
Please define the following
though:
1.
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