See this link: http://www.voip-info.org/tiki-index.php?page=NFAS
- Original Message -
From: "Billy Huddleston" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, October 12, 2005 12:57 AM
Subject: Re: [Asterisk
NFAS isn't needed for redundancy.. You can do this with 2 PRI's just have
them in the same huntgroup at the CO.. All NFAS really does is free up a
extra B Channel.. For 2 PRI's it makes no sense.. For anything over 2 you
pickup 1 extra B channel per PRI.. We use NFAS on our RAS setup for the
I
A SIP phone *could* normally send its media stream directly from phone to
phone, if no transcoding is required, but when using Asterisk the media
stream will always pass through the server, causing a pottential
bottleneck.
So, why not use SER to register all the SIP phones, as it doesn't handle
t
7914 isn't supported with asterisk as of
yet.
- Original Message -
From:
Vasiliy Voropaev
To: [EMAIL PROTECTED]
Sent: Friday, October 15, 2004 6:32
PM
Subject: [Asterisk-Users] Cisco 7960 +
7914 - not worked
I have Cisco 7960 with 7914 operator console.
config info if you want further help
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Billy
Huddleston
Sent: Wednesday, September 08, 2004 11:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco GW and DTM
Sounds like you need to talk to polycom about a reduction in the
capabilities of thier phone after the upgrade and have them move the menu
option back..
- Original Message -
From: "Tor Setane" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: <[EMAIL PROTECTED]>
Sent: Thursday, September
Cisco GW and DTMF problems
What version of IOS 're u using, and what's your dtmfmode in *?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Billy
Huddleston
Sent: Wednesday, September 08, 2004 6:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisc
party. This affects DISA, prompts and
menus.. Has anyone else had this problem?? and use.. I DO have dtmf-relay
rtp-nte toggled in my dial peer..
Thanks, Billy
+--+
| Billy Huddleston Senior Systems Administ
Dude, don't flam.. People don't use HTML capable E-Mail programs, or turn
off html for reason.. Like spam and web-bugs, and/or using classic email
programs like pine and mutt and linux.
Geez..
- Original Message -
From: Karl J. Vesterling
To: [EMAIL PROTECTED]
Sent: Sunday, August 01, 20
Asterisk doesn't support any form of G723 except with Pass through... You
might try G726 or G729
Thanks, Billy
- Original Message -
From: "Arnaud Pignard" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 23, 2004 12:45 PM
Subject: [Asterisk-Users] oh323 & codec G7231A6K3
nks, Billy
+--+
| Billy HuddlestonSenior System Administrator |
| Net-Express http://www.nxs.net |
| 114 Sherway Rd. Voice: 865-691-2011 |
| Knoxville, TN 37922 Fax: 86
Do these work with PRI's as well? What's a ball park price on these?
Thanks, Billy
- Original Message -
From: "Robb Meredeth" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 09, 2004 8:53 PM
Subject: Re: [Asterisk-Users] T1 Hardware Echo Can
> Well, first off we're not u
I've been working with Mark today on fixing this very bug.. The patch
ProgramerTED did may have fixed it, but, I don't think it was the "right"
fix. We should have something done later today on this problem.
Thanks, Billy aka Connor
- Original Message -
From: "Alberto Fernandez" <[EMAI
I failed to mention I'm using a Cisco 2600 with Sip Re-invites.. and YES, I
do have the echo can on the Cisco turned on, The echo is S bad, it's not
even touching it... When we place 1-800 calls or call LD via our offnet
provider, everything works fine, it's just with local calls on the PRI...
I've got the same problem NEAR end echo (We hear the echo on OUR side,
person on the PSTN never hears it..)
We're tyring to get our PRI carrier to run us through a echo can, or
re-write it through a switch they have which has built in echo cans...
Ugg..
Thanks, Billy
- Original Message ---
Yes, I use it. Here's a sample extension of how to use it.
exten => 1234,1,Answer()
exten => 1234,2,MailboxExists(1234)
exten => 1234,3,Dial(SIP/1234,20) ; Try to ring for 20 seconds, no answer
goto voicemail
exten => 1234,4,Voicemail(b1234) ; send to voicemail if busy
exten => 1234,103,Dial(SIP/1
'local' target? What's that?
- Original Message -
From: "Matthew Asham" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 22, 2004 9:55 PM
Subject: Re: [Asterisk-Users] e164.org
> You know, sleep deprivation cause people to do dumb things. The example
> I pasted was hastil
Mark,
Would you please re-config or use a different mail client as to not send
your replies back as attachments??
It's VERY kludgy, and, I'm just going to stop reading them.. along with all
the other folks..
Thanks, Billy
- Original Message -
From: "Mark Elkins" <[EMAIL PROTECTED]>
To: <
That won't work.. That'll DIAL multiple phones/extensions, but will only
bridge 1 of them when it auto-answers..
What we need is a way to have something like meetme call multiple extensions
and bridge them to a meetme confrence (all of them muted but the admin of
course, as it's a one way page) an
hey, can you send me the tone?
- Original Message -
From:
Joe Antkowiak
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Friday, May 07, 2004 4:30 PM
Subject: RE: [Asterisk-Users] Cisco 7940
Phones as paging system?
This is what we have
for this cu
SO, do you have a IDE CDROM?
- Original Message -
From: "Mark Elkins" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, May 07, 2004 4:13 PM
Subject: [Asterisk-Users] 729 licence on scsi
___
Asterisk-Users mailing list
[EMAIL PROTECTED
g711ulaw
- Original Message -
From: "Michael Shuler" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, April 14, 2004 7:01 PM
Subject: [Asterisk-Users] Fax Over VoIP
> Anyone know what protocols support a fax machine i.e. g.729, g.711, etc?
>
> --
s worked up till this point.. What's going on?
Thanks, Billy
+------+
| Billy Huddleston Senior Systems Administrator |
| Net-Express http://www.nxs.net |
I just tried this, and it's not working for me.. I can't call a 2600 or a
CCM... What version of OpenH323 and PWLIB did you all use?
- Original Message -
From: "Marian Durkovic" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, March 18, 2004 10:35 AM
Subject: [Asterisk-Users]
der of codec preference for
> the gateway and see if that fixes your g729 phone and breaks the ulaw
phone
> at the same time.
>
> Alex.
>
> - Original Message -
> From: "Billy Huddleston" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent:
should be using the compatable codec... PLEASE help..
This is going to cause problems in our rollout.
Thanks, Billy
+------+
| Billy Huddleston Senior Systems Administrator |
| Net-Express
unning 3.1, I've got several others, but this one is
the only one that does it...
Thanks, Billy
+----------+
| Billy Huddleston Senior Systems Administrator |
| Net-Express http://www.
he Gateway or
ATA. Is their a way to eliminate this stutter without disabling re-invites?
This is very discontenting to our customers and employees...
Thanks, Billy
+--+
| Billy Huddleston Senior Systems Adm
http://www.nxs.net/cisco_ata_186.htm
- Original Message -
From: "CW_ASN" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, February 08, 2004 12:40 PM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..
> Could you share your 3.0.0 config?
>
> - Original Message
CTED]>
Sent: Sunday, February 08, 2004 12:08 PM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..
> Hi,
>
> Citeren Billy Huddleston <[EMAIL PROTECTED]>:
>
> > Using them with * and a Cisco GW, they're on PUBLIC ip's, No Firewall or
> >
Using them with * and a Cisco GW, they're on PUBLIC ip's, No Firewall or
anything, I am using re-invites. Pretty standard setup. When they lockup,
you can't ping them, or get to the http interface, and I even think the IVR
stops responding when you push the button.
Thanks, Billy
- Original
Anyone had any problems with ATA's running 3.0 software locking up?
Thanks, Billy
+--+
| Billy HuddlestonSenior System Administrator |
| Net-Express http://www.nx
and with the HT-286 you get a Chinaman in a box! :)
- Original Message -
From: "Michael Koehler" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, February 03, 2004 5:45 PM
Subject: Re: [Asterisk-Users] voip phones
> I prefer SIP Phones, Grandstream BT-100 IP-Phone or the HT-28
won't have any problems.. It'll work fine for them.. What can we do to get
this to work like it should?
Thank, Billy
+----------+
| Billy HuddlestonSenior System Administrator |
| Net-Express
IDE/SCSI interfaces, SCSI only installed, WITH IDE CDROM installed with
CDROM in drive. - g729 WILL WORK.
I'm running a system right now with 24 licences.. Tested it with a single
license before purchasing the other 23. You MUST have a CDROM in the drive
when you run the install program.. and MU
ssage -
From: "Senad Jordanovic" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, November 26, 2003 4:14 AM
Subject: RE: [Asterisk-Users] Handytone 286 - calling out
> Billy Huddleston wrote:
> > change dtmf to info on both * and in the handy
change dtmf to info on both * and in the handytone.
- Original Message -
From: "Senad Jordanovic" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 25, 2003 8:01 PM
Subject: [Asterisk-Users] Handytone 286 - calling out
> Hi,
>
> Just received recently released Grandst
loose username=ipphone9
Not needed.. the [109] is really the username
- Original Message -
From: "Anton Yurchenko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, November 24, 2003 11:42 AM
Subject: [Asterisk-Users] strange SIP authentication/authorization behaviour
> When I h
Use CIPE, It's a UDP based VPN solution.
- Original Message -
From: "Alastair Maw" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, November 20, 2003 1:37 PM
Subject: Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
> On 20/11/03 15:44, Chris Hirsch wrote:
> > Hey all..
how could you do this with sip and VOIP?
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 04, 2003 5:40 PM
Subject: Re: [Asterisk-Users] snatching calls
> On Tue, 2003-11-04 at 15:28, Shoval Tomer wrote:
> > Hi,
> >
> > O
;: Found
== Parsing '/etc/asterisk/voicemail.conf': Found
== Spawn extension (default, 9342199, 102) exited non-zero on
'SIP/-0810f210'
Anyone know what all this is about.. The Failed to write frame and the
'multipart/mixed;boundary=uniqueBoundary'
Thanks, Billy
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