Displaying 20 results from an estimated 1000 matches similar to: "TE121B server recommendation"
2009 Apr 09
3
T.38 ATAs
Hello
I am going to try the new Digium Fax for Asterisk product. I'm planning
to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs.
I'm looking at Grandstream HT502 or Linksys SPA2102 ATAs. If anyone has
any experience with these devices, or other recommendations, I would be
grateful if you could share your experiences.
Regards
Ian
2008 Nov 14
4
Looking for a good lightweight Linux softPhone
I used to use IDEFISK, but since it was taken over/renamed into Zoiper
it's been really hard work - now I'm told that they won't support my
chosen distribution - Debian Etch - the current stable version of Debian I
prefer.
So, looking to dump Zoiper and go with something else - I want something
light-weigh (So that rules out Ekiga - and Zoiper was going down the
bloatware route
2009 Jul 06
3
Small site survivability
We are currently moving away from a wide-spread Cisco CallManager deployment
to Asterisk. For many of our small sites we have the routers configured for
what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP
registrar. We are converting to SIP, and from what I can tell Cisco wants a
license for each router to run SRST over SIP...
So my question to the group is: What are
2009 May 11
3
Asterisk w/ Nokia "e" Series Handsets
Anyone using Nokia "E" Series handsets with Asterisk? I'm trying to
deploy some e71's and am having an issue. I can get a single handset
working, but when I try to create a SIP profile on the second phone, it
won't allow me to save the profile, saying that devices in the same
"realm" must have identical username and password.
Anyone have a workaround for this
2008 Aug 30
1
Heist of MagicJack SIP credentials?
While I myself am not a MagicJack user, I'm curious as to whether anyone
here has managed to heist their MagicJack account's sip credentials, and
use them to terminate calls using asterisk. Apparently it's as simple
as sniffing the SIP credentials. If so, said person would enjoy
unlimited termination for $20 year while retaining the flexibility of
setting their CallerID to a
2009 May 26
2
Converting Cisco 7961 to SIP
As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files:
apps41.8-4-3-16.sbn
cnu41.8-4-3-16.sbn
cvm41sip.8-4-3-16.sbn
dsp41.8-4-3-16.sbn
jar41sip.8-4-3-16.sbn
2009 Mar 16
8
Good phone near $125
I was looking at the aastra 9133i, however I was informed that this phone is
no longer supported. What are good phones around the $100 - $125 price
point? (Need POE)
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200 david at safedatausa.com
2008 Nov 07
3
TE121B Doesn't Fit PCI-E Slot
I have 2 HP Proliant 365 G5 servers with PCI-E risers. I bought a Digium TE121B single port card. When installing the card, the slot on the card doesn't quite line up with the tab in the PCI-E slot. If I loosen the front plate on the card, Ican sort of make it plug in, however, the card won't go in far enough to screw down the plate. I tried the card in the other server and had the same
2009 Aug 17
0
Echo on TE121B with hardware echo module
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
I recently upgraded my Asterisk system to Dahdi and now I have an echo
problem.<br>
<br>
I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium
TE121B PCI express card with a HARDWARE echo
2007 Aug 27
3
OT: DELL Platforms
Hello list,
I have a customer who is interested in standardizing on dell servers for
asterisk deployments.
Has anyone had success with a particular configuration?
Anything specifically to watch out for?
Thank you for your time,
Art
Arthur Miller
Sr. Sales Associate
VoIP Supply, LLC.
454 Sonwil Drive
Buffalo, NY 14225
716-250-3871 OFFICE
716-630-1548 FAX
arthur at
2009 Aug 06
1
OT - Opensourcesip.org
Anyone have any firsthand experience implementing OpenSBC
(opensourcesip.org)? Have a possible consulting gig referral.
Cory J. Andrews
Director New Market Initiatives
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
candrews at sayersmedia.com <mailto:brett at voipsupply.com>
Have I
2008 Jul 24
2
Audiocodes MP-11X configuration to work with Asterisk
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It
registers fine and I can call between the MP-114 and other extensions,
but I'm not having much luck with the FXO ports. syslog shows the
problem to be in the MP-114 configuration.
Can anyone help?
2008 Dec 19
3
Pre-routing manipulation of calls
This is concerning an Asterisk 1.4.18 server.
We have approximately 70 DID numbers. Incoming calls are placed into
the "incoming" context (by zapata.conf) and are routed based on the
dialed number.
I want to do some manipulation (CallerID name override) to all incoming
calls before they are routed. I would prefer to avoid duplicating the
necessary code in each DID extension stanza,
2007 Aug 28
9
Dell SC1430 + Digium TE110P = Digital Noise in PRI
Hi list,
I have a terrible noise issue with Dell SC1430 + Digium TE110P. The
digium card is not sharing interrupts with any other device, as I saw in
Dell's BIOS and also with "lspci -vb" command.
After changing coax wire, UTP, balum, digium card ... I have found that
the problem is in Dell box, so now I'm running the same Asterisk config
in other server with the same
2004 Dec 08
7
sangoma
Good day all
Is there someone that's got asterisk working well with a A101/E1 card
Apparently they don't have RBS support?
Please advice
Thanks
Altus
2008 Oct 13
1
Tracking T1/PRI channel status - inbound vs. outbound
I need to monitor the states of my T1/PRI Zap channels. Specifically, I
need to be able to programmatically determine whether a channel is
unused, carrying an inbound call, or carrying an outbound call.
Using the manager interface, I can easily tell whether a Zap channel is
used or not by looking at the results of:
Action: Command
Command: zap show channel <x>
Or:
Action:
2009 Aug 25
2
Echo
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
I recently upgraded my Asterisk system to Dahdi and now I have an echo
<br>
problem.
<br>
<br>
I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium
<br>
TE121B PCI express
2004 Aug 06
5
bandwidth negotiation
Does Icecast support bandwidth negotiation like Real's server? If so, how
would one configure this (can't find it in the docs or list archives).
If not, is there any interest in adding this capability?
--
Kevin DeGraaf
--- >8 ----
List archives: http://www.xiph.org/archives/
icecast project homepage: http://www.icecast.org/
To unsubscribe from this list, send a message to
2008 Jul 31
1
PDC cannot become master browser; cannot change passwords
I am having two problems, possibly related, while performing
pre-deployment testing of a Samba/OpenLDAP PDC with data that was
vampired from an NT4 PDC. The Samba server fails to become a local
master browser, and password change attempts (from a Windows client) fail.
I followed Samba-Guide/ntmigration.html (taking some liberties with
various items of configuration), ending with step #19.
2006 Feb 23
9
Linksys WIP300 WiFi Phone
Whoo hoo! I just received my WIP300 from voipsupply. I have to let it
charge before I can play with it.
A few quick comments:
- I started a Wiki page at voip-info to post issues, firmware news, etc.
I really like the wealth of info on the GXP-2000 page, so I wanted to
start something similar for this phone.
http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300
- My kit