Wehave (2) cartons of (56) AC Power Cubes for
the Cisco 7905, 7910, 7940 and 7960 IP Phones.
These are brand new, and include the power
cord.
They come with a 1 year warranty.
Cost is $17/ea, minimum order of 10
pcs.
Cory
Andrews***b2 Technologies454
Curious if anyone has any feedback on Nufone
voip pbx.
Cory J Andrews
**
b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
**
866.44.B2TECH X22
local 716.630.1555 X22
fax 716.630.1548
***
[EMAIL PROTECTED]
web
My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of
service. They were deployed for about 6 months. These include the AC power
adapter and station license. We also have some other related equipment. If
someone is reading this and is interested, shoot me an email
[EMAIL
Can anyone point me to some online documentation showing how to reset a
CP-7960 to factory default settings. I have some that are configured for
Callmanager and I want to get them back to generic default config. Any info
is appreciated.
Thanks
Cory Andrews
Never used in production - $3750/ea Email [EMAIL PROTECTED] if interested.
Cory Andrews
*
b2 Technologies
*
voice: 866-44-B2TECH X22
fax: 716.630.1548
email: [EMAIL PROTECTED]
___
Asterisk-Users mailing list
Hello everyone,
I'm hoping someone can help me with this. I have a business customer in
the U.S. (Michigan, ATT Territory).
I need to get 4 trunks into an asterisk Box. My intention is to use an
Eicon Diva Server card with 2 BRI Circuits. The reason for this is that
the business needs DID's
We have (10) Dialogic D/240SC-T1 REV 2 voice boards. Prefer single buyer,
take all (10) for $675/ea
Cory Andrews
*
b2 Technologies
454 Sonwill Drive
Buffalo, NY 14225
*
voice: 866-44-B2TECH X22
fax: 716.630.1548
email: [EMAIL PROTECTED]
Have (200) Brand New power cubes (AC Power Adapter
with AC Cord) - compatible with Cisco CP-7910, CP-7940, CP-7960 and equivalent
"G" models.
$25/ea - Minimum Purchase (10) Units.
Email [EMAIL PROTECTED] if
interested.
Regards
Cory
Andrews***b2
For high-availability Asterisk when using PRI Lines, you might also want
to check out our product (FSV-4PFS). It's available at
www.failsafevoip.com
Bill
Also to add to the last post does this device have hardware echo cancelation?
if it does it could be a great replacement, if not may not be
Dear friends,
As a community service, FailSafeVoip is providing a free US Based Echo
Test. The service is running on a high performance asterisk box and is
connected via a fully TDM T1-PRI. The test server is based in Michigan.
The test extension is written simply as:
s,1,Answer
s,2,Echo
I have been a reseller subscriber of pipecall since
they started, however I am really struggling to get pipecall to work for
outbound or inbound calls. I get errors that the registration has timed out.
I have tried many variations of the register command
register = [EMAIL
Sorry about this, I have been struggling with the basics of my
asterisk config.
I set up two sip peers and two phones. And I set up lots of dial
masks for outgoing calls, all my outgoing calls were working great, however
incoming calls were a different matter altogether, I cannot get
Then there is no way to make asterisk listen on multiple port ? Currently
iptables 5091 forward to 5060 is working but this should really be a
asterisk feature :(
On 16/12/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Mail list wrote:
Yes i read that on voip-info wiki but i have
Can anyone point me to where I might obtain the SIP 3.0 image for the
ATA-186 Analog adapter. I'm willing to pay for it. I have a Cisco login
but am apparently not authorized for this, just trying to get my fax working
with asterisk and I need SIP 3.0. Any advise appreciate.
Thanks
Cory
How do you configure extensions.conf to let you punch out to
VoicemailMain when an individual voicemail prompt has picked up?
We have a few extensions set up. voice mail is extension 8500 and we
have another extension for SIP on extension 12. SIP dials out fine.
SIP can dial 8500 and get
Could you have asterisk running and not allowing you to overwrite while
trying to install? Do you have root rights to create files in the
asterisk folders?
John Chambers wrote:
After doing cvs checkout -r v1-0_stable asterisk and typing the
usual make clean ; make install, I got these
Condition: New Open Box
Warranty: 90 Days
Cost - $130/ea Minimum Order 5pcs
Contact [EMAIL PROTECTED] for details
Cory Andrews
++
b2 technologies
454 Sonwil Drive
Buffalo, NY 14225
++
email - [EMAIL PROTECTED]
voice - 716.630.1555 X22
fax - 716.630.1548
web -
Try www.VOIPSupply.com they have the model 300, 500 and 600 phones
available.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker
Sent: Monday, October 18, 2004 5:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hello list,
Is there any way to use a2billing without the IVR for the sip/iax users. (authentication is done by the user id and pass as user registers with asterisk).
I want to dial the destination number to the asterisk. for example:
user dials,
exten =_011.,1,DeadAGI(a2billing)
system will
Hello list,
Is there any way to use a2billing without the IVR for the sip/iax users. (authentication is done by the user id and pass as user registers with asterisk).
I want to dial the destination number to the asterisk. for example:
user dials,
exten =_011.,1,DeadAGI(a2billing)
system will
hello list,
i have a dial plan
exten =12345,1,Dial(sip/100sip/101sip/102,30,rt)
so when calls come to 12345 all the phones 100, 101,102 rings.
if any person receives the call it connects the call with no problem. but
in the CDR it does not show who receied the call.
it shows
hello list,
in my asterisk i have blind transfer and attendent transfer.
when call Z which is a public call through Capi(BRI) is received by user A he can see the Caller ID of Z and
if user A blind transfer the call to user B, user B can see the caller ID of user Z but
when user A attendent
hello everybody,
i have updated my rpm asterisk to current cvs 1.0.9. I had been usingrpm asterisk which comeswith suse 9.2. the main reason i updated my asterisk to get the attended transfer feature. i haveinstalled anothercvs
1.0.9 asterisk in Redhat 9 and it works perfect.
here what i found:
hello everybody,
i used asterisk with 2 ISDN BRI AVM cards in paralel with a panasonic ISDN pbx for testing putpose.
is this also possible to use E1PRI to use in parallel with a simence PRI pbx for test purpose?
|---asterisk
public line|
|---PBX
best regards
shaon
hello asterisk users,
i an using asterisk cvs 1.0.9 in a pIII 733mhz 256MB RAMredhat 9.
i have a TDM400P with 2FXO and 2FXS modules. in my fxs i want to get australian dial tone and for all asterisk operation i want to use australian tones. by default it is US. to change this i have edited
hello list,
ineed to setup an asterisk system with 5 ISDN trunks. ifound C4 cards but they are very expensive.i found that if i use 5 AVM Fritz! cards itwould bevery cheap. i want to use 2 boxes. 3 in boxA +2 in boxB=5 isdn.
and i want,this two boxs to work as a single box so that one box can
Hi folks,
I have a E1 interface and i need to get some informations about it,
like:
Informations about the Layer 2 (state - active, inactive)
HDLC messages (time, channel, events ...)
Messages from Layer 2 and 3 (Rec. Q.921 and Q.931)
Number of sent messages and bytes, received
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