Displaying 20 results from an estimated 2000 matches similar to: "opinions on oneunified.net as asterisk provider"
2003 Oct 29
3
FW: Voice/Data mixed routing over Digium E1/T1 Card
> The documentation mentions that the Digium channels can be split into some
> voice channels and the remainder of the channels used for routing IP
> traffic.
>
> Does any one have this in use in conjunction with Asterisk? Does it work
> well? Would you recommend it for a production server?
>
> Obviously, if this works, this makes for a cost effective platform where
2003 Nov 18
2
ISDN Card Types for Europe
What types of ISDN BRI cards work well in Europe (Guadeloupe, Martinique and
France) ? For example: AVM C2 or AVM C4 or eicon Diva server 4 BRI? Any
others? Which driver is appropriate?
Ray Burkholder
ray@oneunified.net
http://www.oneunified.net
704 576 5101
--
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2004 Jan 12
2
SIP-Client for Handheld PC
Anyone know a sip-client that will work on a Handheld PC running WINCE for
HPC.
I can find some for PocketPC, but the wont work on my HPC
??
/HHA
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2003 Nov 01
2
Making a Skinny phone talk to Asterisk
I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm
a little unsure as to how get the phone to figure out which ip address it
should register with when it boots.
How do I do that?
I already have a tftp server for my SIP based phones. Do I need a tftp
server for skinny configs at all? And if so, can it be the same tftp server
as the SIP ones use (I'm not sure
2004 Jan 05
1
Identifying the Originating Cisco SIP Gateway
I have several Cisco SIP gateways sending calls to Asterisk. Because the
gateways don't have user-agents, they don't authenticate with Asterisk. And
because they don't authenticate, they use the default context in the
sip.conf file.
Is there a way to either:
A) identify the inbound gateway with a variable, in channel info, or the
manager interface? If there was a ${SIPDOMAIN} for
2003 Aug 25
1
Intercom with Cisco SIP 796x phones?
I read about this intercom stuff on page 62 & 63 of the book "Developing
Cisco IP Phone Services" isbn 1-58705-060-9. Primary calls take place
on streaming channel 0. When streaming channel 0 is not in use,
streaming channel 1 can be used for asynchronously streaming (in and
out) stuff like voicemail, email, and, yep the one we want, intercom.
Page 87-88 of the book talks about
2003 Dec 31
3
Java?
We needed the client browser to be open all the time for dynamic data to
load without the page refreshing. After looking at all of our options we
decided on programming it ourselves using flash rather than java.
We have a flash frontend thats tied to our backend mysql DB. We use it
for loading web site traffic data, email opens, click-throughs,
bouncebacks, stats, etc. It could also be used with
2003 Nov 12
2
Canadian VoIP termination?
Hi,
Does anyone know of Canadian VoIP termination providers? I have
Canadian customers and would like to provide Canadian dial in and dial
out (canadian callerid).
Thanks!
2004 Jan 16
11
Remote reload Cisco 7960
Does anyone have a working way of having a Cisco 7960 reload its config
remotely. I have tried some of the scripts that I have found on the web,
but to no avail. Thanks for the help.
B. J.
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2003 Dec 23
1
OT: SIP vs. Skinny protocol
I assume there are several people on this list that
have Cisco Call Manager implementations under their
belt....
We are beginning a call manager implementation and
the first question I asked Cisco was, should we use
SIP or Skinny. Cisco is pushing me towards Skinny,
saying that I will lose some functionality with SIP.
They also say that most of their customers implement
skinny.
I see two
2004 Jan 13
1
cisco 7910 phone
Hi All
Will cisco 7910 ip phone compatible with Asterisk? I know that 7960 are
fine.
David Kwok
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2004 Jan 06
1
IAX2 Trunk two Asterisk boxes.
I need to get 2 Asterisk servers working together. I have been reading
and doing just about every example I have been able to find here on the
list and the Wiki. It's now gotten to the point that nothing on box2
seems to be working. I seem to have a major problem understanding the
format. Here is what I have so far. It's 3 days of hair pulling and
nothing seems to work!
Asterisk box 1
2004 Jul 20
4
Wireless SIP Phones
Hello,
I found serveral discussions about the Zyxel ePhone Prestige P2000W and
the WiSip from Pulver Innovations on this mailings list but still have some
questions:
1) are there other affordable wireless SIP Phones on the market? I haven't
seen or found anything else till now ...
2) is p2000w and wisip the same hardware?? so could I use firmware
from both companies regardless of what
2003 Oct 26
5
Extensions Problem
Hello again,
Here's the next big issue, I thought I'd let you munch on. We are utilizing
Cisco 7960's and the following entries in our extensions.conf file:
Exten => 1637,1,Dial(SIP/100)
Exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo)
Exten => _NXXXXXXXXX,2,Congestion
Exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo)
Exten => _1NXXXXXXXXX,2,Congestion
These
2004 Jun 16
6
Invalid Extensions -- More like traditional PBX systems?
I was wondering if there was a way of setting up the dialplan in a way
that if you dial an extension that is NOT in the dialplan then it would
play a not-in-service gsm file and then play congestion tones. I would
rather like this better than just hearing a busy signal on my phones.. I
DID search around on the wiki and using google and could not find anything.
Thanks.
--
Stephen Rosebush,
2004 Dec 07
0
Comdial PBX -- can use Asterisk as VM
If your vmail is connected via serial to your PBX it is most likely using
SMDI for MWI which isn't supported by Asterisk. I seem to remember that
this was submitted as a feature request with a bounty tied to it.
Keith
Date: Tue, 7 Dec 2004 12:58:11 -0500
From: George Herndon <gherndon@eyeontech.com>
Subject: Re: [Asterisk-Users] Comdial PBX -- can use Asterisk as VM
2004 Jan 08
3
Asterisk success stories in small-mediumoffice environments?
I'm not really looking for working configurations as much as I am
looking for people who can say "This is a solid product and I trust my
business to a solution running Asterisk."
As far as pre-sales work... Well, tell that to my consultant.
I'm quite excited about *. I've got my company sold on it, they just
want some reassurance that it's ready for prime-time
2003 Aug 25
6
SIP vs SCCP vs XML
>
> No, this is not the case currently with any of the Cisco SIP software
> loads that I am aware of. If you find this to be incorrect, please
> let the list know. Cisco has not deployed much of the featureset in
> their SCCP phones (such as paging/intercom) into the SIP phones due
> to lack of standards/interest/political capital.
>
> JT
Ok, after further
2003 Oct 27
14
Answering Machine Detection
Does anyone have any recommendations on implementing Answering Machine detection for call generation programs?
What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example:
If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,
2003 Aug 20
14
Is Asterisk ready for "real" use?
Okay,
I am facing a move in two months to newly renovated space. I
have to decide *this week* between:
A) Pull LAN and phone cables, prepare to move and expand our
"traditional" PBX (Panasonic KX-TD1232 and VPS200).
or
B) Pull only LAN cables, go VoIP, use Asterisk as PBX.
It is *not* an option to purchase a VoIP system package from
Cisco, 3com, etc. Installers are getting an