David Uzzell wrote:
It would be great if you could share with the rest of us newbie type
people some of your extensions.conf and iax.conf to do things especially
like the last one were you can dial in and pin and make long distance
calls. This does very much intrest me especially :)
Cheers
Try
this e-learning tutorial. It requires macromedia flash.
http://www.cisco.com/warp/public/779/largeent/avvid/products/7960/router_page.htm
http://www.cisco.com/warp/public/779/largeent/avvid/products/7940/index_1020.htm
Regards,
Walid
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Good day all
We have Grand Stream BT-100 phones
The transfer button work well, for blind transfer
What the users want to do is, when a call comes in and asked to be
transferred to another extension,for example 100,they 1ste want to speak
to exten 100,then have the option transfer or not to
/proc/sys/fs/file-max
This file defines a system-wide limit on the number of open
files for all processes. (See also setrlimit(2), which can be
used by a process to set the per-process limit, RLIMIT_NOFILE,
on the number of files it may open.) If you get
Hello!
I see many of you experience troubles with H323 stack. I am focusing
on building H323-SIP Asterisk based softswitch with all codecs
supported (including G729 and G723).
I can setup Asterisk from scratch with H323 support or solve your h323
nightmare with existing asterisk system for for
Hi list,
I have been folowing the SS7 for * thread and it got me wondering about the
current status of SS7 for *.
Anybody knows if ISUP going to be supported?
Yours,
Hadi
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.290 / Virus Database: 265.4.5 -
To answer my own message, I need to set the REDGNO (0x74) number to
the
originating number in the PRI SETUP. Example can be found here:
http://pastebin.ca/2783
Does anyone know how I can set this with asterisk?
I have only look quickly at the code, but it seems as if asterisk will
copy whatever
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
well
running GSM or some fancy codec over GPRS or UMTS may well make sense :)
roy
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asterisk h323 wrote:
Hello!
I see many of you experience troubles with H323 stack. I am focusing
on building H323-SIP Asterisk based softswitch with all codecs
supported (including G729 and G723).
I can setup Asterisk from scratch with H323 support or solve your h323
nightmare with existing
Hi all,
When I transfer a caller to another internal extension that is either
off the hook or in use, it immediately drops the caller. Is there a way
that I can put something in maybe my extensions.conf to transfer that
caller back to the original extension if the called one is busy?
On Thu, 9 Dec 2004, Roy Sigurd Karlsbakk wrote:
I have only look quickly at the code, but it seems as if asterisk will
copy whatever is in the channel variable cid.cid_rdnis for the calling
channel to the outgoing channel in app_dial unless the channel is set
to
forward calls.
is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I'm sure most people are aware of the ability of pppd to answer calls
coming in from a standard serial modem (or at least that is the way
I
understand it to work), authenticate the user and issue it an IP
address. With the proper ip
On Thu, 9 Dec 2004, el Flynn wrote:
asterisk h323 wrote:
Hello!
I see many of you experience troubles with H323 stack. I am focusing
on building H323-SIP Asterisk based softswitch with all codecs
supported (including G729 and G723).
I can setup Asterisk from scratch with H323 support or solve your
You need firmware 1.0.5.16 (Broken message button for voicemail) or 1.0.5.18
(Still in Beta, phone display '403' error about once per hour for 10 seconds
or so. In order to use attended transfer you place the caller on hold by
pressing the flash button and then dial the third person. Once you
Ryan Sackler wrote:
I'm sure most people are aware of the ability of pppd to answer calls
coming in from a standard serial modem (or at least that is the way
I
understand it to work), authenticate the user and issue it an IP
address. With the proper ip forwarding/masquerading techniques, this
can
On Thu, 2004-12-09 at 10:56 +0800, TinKoon wrote:
Hi,
I was told the Carrier Access Adit 600 supports ethernet based channel
bank right out of the box. But I cannot confirm whether this is true as
nobody seems to use the Adit 600 this way.
While the Adit 600 has a ethernet port standard,
Look at AudioCodes (MP-108, MP-124)
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 09, 2004 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Ethernet Channel Bank idea
On Thu,
I would like to try and use a surplus (decommissioned) Cisco AS5350 with
Asterisk. Bascially the 5350 will connect to the PRIs and send the calls
to asterisk (and likewise for calls from Asterisk to the PSTN). The 5350
has both PRIs and DSPs, so it should be suitable.
Has anyone done this
Title: Message
Hi
all,
I had
a chance to use some call conferencesthat had some very neat
functionalities:
- When
you call you are first asked for your name
- When
someone joins the conference a message "name is now joining the
conference." is played.
- When
someone leaves the room a
Hi,
At the * console I periodically get these messages:
Dec 9 10:58:11 WARNING[-1248765008]: chan_iax2.c:5021 socket_read: midget
packet received (1 of 4 min)
Which seem pretty inocuous.
Google say (almost) nothing about that subjet.
What does it mean?
--
Field Artillery lends dignity to
Hi,
I have the exact same problem, we have two Eicon DIVA Cards (BRI UK),
using chan_capi by Junghann.
The cards have been tested and work perfectly, if we make two outgoing
calls simultaneously, and someone calls us, they get a busy tone or call
failed, yet capi info says 2 channels are still
On Thu, 2004-12-09 at 02:49, David Uzzell wrote:
Jean-Michel Hiver wrote:
I've been setting * at home just to train myself with it. Here is what I
have:
- IVR menu
- music on hold / transfer
- voicemail
- transparent Zap or IAX routing
- I can call home, dial a pin and make long
On Thu, 9 Dec 2004 10:45:08 +0200, Hadi Jadallah [EMAIL PROTECTED] wrote:
Hi list,
I have been folowing the SS7 for * thread and it got me wondering about the
current status of SS7 for *.
Anybody knows if ISUP going to be supported?
Hadi,
at this stage ISUP is the only User/Application
Go seacre for asterisk tiptricks
Ronan Eckelberry writes:
Hi all,
When I transfer a caller to another internal extension that is either
off the hook or in use, it immediately drops the caller. Is there a way
that I can put something in maybe my extensions.conf to transfer that
caller back
Hi All,
I have just this minute found a solution that works
for us. The problem is not with the asterisk
configuration but with the configuration of the Eicon
Card.
I use the Eicon-supplied http server on port 10005 to
configure the Eicon card. On the hardware
configuration page, set:
CAPI Call
Thanks for the info, unfortunately that still doesn't work for me.
Making two outgoing using ISDN.
Contr1: 2 B channels total, 0 B channels free.
Contr2: 2 B channels total, 2 B channels free.
*CLI capi info
Contr1: 2 B channels total, 0 B channels free.
Contr2: 2 B channels total, 2 B channels
Hunt group is set to Standard Operation (default)
If you have 2 controllers presumably you have 2 ISDN2e
lines. Have you asked BT to set up a hunt group so
that the same number can be used to dial in to any of
your 4 available channels ?
Best wishes,
John
--- Craig Waddington [EMAIL
On Thu, Dec 09, 2004 at 12:01:54AM +0200, Shoval Tomer wrote:
Has anyone had successfully installed more then two digium wildcards in
the same machine?
I'm going for four.
As others have said, you need to make sure you aren't sharing IRQs with the
Digium cards. One way to easily avoid it is
Hello,
I am getting this kind of Warning in the Asterisk console, but i don't
know why.
WARNING[8200]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED] for seqno 102
(Non-critical Request)
Could you give some clue to solve this problem?
Thanks in advice.
Ismael.
We've been using Broadvoice with * for several months, mostly successfully.
However, sometime last weekend (Sat, Dec 4 - Mon, Dec 6), BV seems to
have stopped passing DTMF on incoming calls. We've not made any changes
to the * system during this time. We even switched to using their DC server
Hi *'s,
Back Again
I want to use PostgreSQL instead of MySQL basically i want to create an
application (calling card),what is the procedure i mean in which files
i saw several config files and change it slightly but not sure about it
i search wiki alot but on wiki almost all info about MySQL
Try this:
http://lists.digium.com/pipermail/asterisk-users/2004-March/039819.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ismaelg
Sent: 09 December 2004 12:07
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] A waning console error
Hello,
I
Answering some questions...
On Tue, Nov 30, 2004 at 10:28:13AM +1100, David Uzzell wrote:
Michael Graves wrote:
On Mon, 29 Nov 2004 10:09:26 +0200, Gilad Ben-Yossef wrote:
Alex Brecher wrote:
Which Distro is the most commonly used distro with Asterisk please ?
I don't know which is
Hi
Version 0.9.0 of Xorcom Rapid Debian/GNU/Linux/Asterisk has just been
released.
Main changes:
* A decent version of Asterisk/Zaptel (1.0.2) is provided
* Includes a better default configuration
* Automatic detection of the most common Zaptel cards
* Contains more optional software (apache,
Hi,
i am trying to connect to freenet.de from an asterisk server behind a nat
firewall.
Asterisk couuld register to freenet, but i get an error :
-- Executing Dial(IAX2/[EMAIL PROTECTED]/5,
SIP/[EMAIL PROTECTED]|45|r) in new stack
-- parse_srv: SRV mapped to host iphone.freenet.de,
On Thu, 9 Dec 2004, Stojan Sljivic - Pamet wrote:
I had a chance to use some call conferences that had some very neat
functionalities:
- When you call you are first asked for your name
- When someone joins the conference a message name is now joining the
conference. is played.
- When
Hi.
Please excuse me asking this again. But it really puzzles me.
We're installing asterisk at a branch office at NJ (HQ is at
Petach-Tikva)
It'll need to support 5 POTS lines, 11 analog extensions and four VOIP
phones.
I wanted to go with a T1 card from digium and a channel bank, but we
have a
On Wed, 08 Dec 2004 22:32:16 -0600, Andrew Aken [EMAIL PROTECTED] wrote:
Does anyone have any experience with running asterisk on multi-processor
computers (dual or quad)? Does asterisk on the latest Linux distros take
advantage of the extra processors, or does it predominately utilize a
On Thu, 2004-12-09 at 14:23 +0200, Shoval Tomer wrote:
Hi.
Please excuse me asking this again. But it really puzzles me.
We're installing asterisk at a branch office at NJ (HQ is at
Petach-Tikva)
It'll need to support 5 POTS lines, 11 analog extensions and four VOIP
phones.
I wanted to
Sorry to ask such a basic question:
I have a * box with 2 nics in the following setup:
Internet
|
192.168.5.253 (firewall)
|
192.168.5.xxx network (gw 192.168.5.253)
|
192.168.5.10 (* nic 1)
192.168.6.10 (* nic 2)
|
192.168.6.xxx network
The netmask for both networks is 255.255.255.0
The
I'm trying to use an hfc based pci card with asterisk but every call fails
falling in the congestion extension.
exten = _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr)
exten = _0.,2,Congestion
Looking in the syslog i can see:
isdn: HiSax,ch0 cause: E001B
it seems that this is a terrible error
On Thu, Dec 09, 2004 at 07:36:41AM -0500, Tony Nichols wrote:
On Wed, 08 Dec 2004 22:32:16 -0600, Andrew Aken [EMAIL PROTECTED] wrote:
Does anyone have any experience with running asterisk on multi-processor
computers (dual or quad)? Does asterisk on the latest Linux distros take
advantage
I just set up a Polycom 500 on *. Every few calls I make, the call
connects and the receiving party can hear me (thru Broadvoice), but I
still get ringing on my end, as if they never picked up. * logs look
just fine. Does any one have any suggestions? Thanks.
So I thought of installing a combination of four pci cards in the
machine, and everybody on the list just keeps telling me it won't work.
You have 5 POTS lines and 4 X100P cards? Sounds like a complete drag...
At any rate, why don't you buy a TDM400P with 4 FXO ports? I've bought
one off
You need to have asterisk route these calls.
You need to point the phones to it as their default gateway, and the
pc's need to point to it as the gateway for the .5 network.
Explaining how it's done is very off list.
Please contact me off list if you want any pointers.
-Original
Hi,
can i configure a different outbound proxy for each sip-peer?
Bye
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To UNSUBSCRIBE or update options visit:
Asterisk does not do anything in this vein.
Simply
% echo somevalue /proc/sys/fs/file-max
a good starting point for this value would be double your existing
value.
% cat /proc/sys/fs/file-nr
will give you your existing max files. I would also suggest doubling
your inodes as well.
%
Hello,
one of the numbers where historically configured to act the following way:
123456: Ring All Desks
123456-1: Ring Desk 1
123456-2: Ring Desk 2
... (I think you get the idea)
Configuring asterisk to do the same isn't that hard, but I now have one
problem, with users calling that number from
On Thu, 9 Dec 2004 10:45:08 +0200, Hadi Jadallah [EMAIL PROTECTED] wrote:
Hi list,
I have been folowing the SS7 for * thread and it got me wondering about the
current status of SS7 for *.
Anybody knows if ISUP going to be supported?
Hadi,
at this stage ISUP is the only
On Wed, Dec 08, 2004 at 08:43:10PM -0600, nik martin said:
Anyone ever thought about an Ethernet based channel bank? Basically a
rack mount set of 24 IAXys? That would be cool, IMO. No wrangling with
zaptel, etc. IAX as the * - Channel bank protocol.
Yes. Search the list :-)
My idea
As I am not a Polycom dealer, I cannot download the software from their
site. Any alternative locations you know of? Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tor Setane
Sent: Thursday, December 09, 2004 7:47 AM
To: Asterisk Users Mailing
Hi,
I'm using an Adit 600 Channel Bank with *. I love it and it works
really great for my FXS lines. One problem that I have with it (It's
really not a problem yet, but it's a potential one) is that I've
scoured the manaual for the Adit to see if there's a way to dump out a
config file from the
is this related to the REDGNO header in PRI?
It seems to be. Libpri fills in the fields (redirectingnum,
redirectingplan, redirectingpres, redirectingreason) in the libpri call
structure when Q931_REDIRECTING_NUMBER (0x74) is received. Similarily,
if
they are set on calling out that IE is sent.
I think its print config
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Stewart
Sent: Thursday, December 09, 2004 8:27 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] [OT] Adit 600 Question
Hi,
I'm using an Adit 600 Channel Bank with *. I
On Tue, Dec 07, 2004 at 08:31:27PM +, Corvin wrote:
I've compiled chan_capi - but I can't force it to work.
Problem description? Error Messages?
If you want to use chan_capi, it has to be TE mode. NT mode is not
possible.
Hfc based ISDN cards will generate lots of interrupts.
nik martin wrote:
Anyone ever thought about an Ethernet based channel bank? Basically a
rack mount set of 24 IAXys? That would be cool, IMO. No wrangling with
zaptel, etc. IAX as the * - Channel bank protocol.
Just an idea...
Allied Telesyn VoIP Access Device
On Thu, 9 Dec 2004 13:18:15 +0100 (CET), Peter Svensson
[EMAIL PROTECTED] wrote:
Another way is to do this through the dialplan. The steps would roughly
be:
1. Answer the call
2. Authenticate the user using authenticate, dialplan logic or an
AGI script/program.
3. Play a message
Has anyone used the Swissvoice IP 10S (www.swissvoice.net) VoIP Phone with *?
Adrian
--
Adrian Walker
[EMAIL PROTECTED]
===
This email has been scanned for Virus infection by MessageLabs
For more information please
Hi Sean,
Thanks for your reply, but that wasn't exactly what I was getting at.
I don't need to increase the system's imposed limit on the number of
open files. I'm more concerned to see if anyone has run across a
memory or fd leak in asterisk that sucks them all up.
There should be no reason
Hello everyone,
Since this seems to keep coming up, I added an entry to the Wiki last
night:
http://www.voip-info.org/tiki-index.php?page=Asterisk+hardware+interrupts
It has been able to clear things up in the past. (By past I mean
yesterday, which was almost the exact same thread).
--
I have a problem with incoming
calls being recorded after a supervised transfer.
Call comes in, receptionist
answers, caller put on hold, Asterisk Monitor is recording, caller is on Hold,
Callee picks up the call, Asterisk Monitor Stops.
All recorded calls are named CallerID
to
Good day all
I have asterisk running on a public ip and a client running behind a natting
firewall with a Grandstream bt-100.Is there something special I should do to
get it working,I got other users working using host=dynamic in sip.conf
Please advice
Thanks
Altus
Hi Dean -
Noah, what client were you using on your treo for this 600ms voip call?
Oh, I wasn't using a SIP client (is there one for palm?). Sorry if
that was misleading - this is just web browsing and email. Once the
connection gets going, it is able to do the 2.2 KB/s that standard GPRS
I have experienced this, but on an intermittent level. I didn't change
anything, but now when I call with my Cingular Cell phone, my IVR doesn't
accept any digits I press. I thought it was completely shot until I called
with my Verizon cell phone, the IVR recognized my digits. It seems to be
like
On a new * asterisk install onto new install Gentoo 2003.4 upon startup
of asterisk:
WARNING[16384]: loader.c:248 ast_load_resource:
/usr/lib/asterisk/modules/res_perl.so: undefined symbol: PL_thr_key
WARNING[16384]: loader.c:429 load_modules: Loading module res_perl.so
failed!
perl -v = v5.8.5
-Original Message-
From: Adrian Walker [mailto:[EMAIL PROTECTED]
Sent: 09 December 2004 14:21
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Swissvoice IP 10S VoIP Telephone
Has anyone used the Swissvoice IP 10S (www.swissvoice.net)
VoIP Phone with *?
Hi,
-Original Message-
Has anyone used the Swissvoice IP 10S (www.swissvoice.net)
VoIP Phone with *?
Yes.
Florian
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On December 8, 2004 09:56 pm, TinKoon wrote:
I was told the Carrier Access Adit 600 supports ethernet based channel
bank right out of the box. But I cannot confirm whether this is true as
nobody seems to use the Adit 600 this way.
I use Adit600s and have not seen anything that would suggest
What sells our clients in our Demo room is that we have a Cisco 7960/7940,
Polycom IP500, IP600, a GrandStream BudgeTone 101 and of course a laptop
with FireFly on it. They love to see how all the different phones can
integrate. It shows them that they're not locked down to one model phone
always,
Dell server ( 2 x PCI-X, 2 x PCI-64, 2 x PCI-32), Xeon
2.4GHz, 256MB RAM, 80GB IDE disk). Currently at 99 quid (+VAT
+ 50quid shipping).
Cheers for the heads-up, Steve, myself and a colleague have ordered two
of these boxes each. When a second colleague went to order, they've sold
out.
Alex Barnes wrote:
http://www.definitive-edge.com/index-2Swis.htm
This would be interesting except it appears to be a bit pricey.
Am looking for a nice quality SIP phone that supports Message Waiting
Indicator (Grandstream are too Fisherprice for my liking).
If anyone has experience of it and also
On December 8, 2004 05:12 pm, Paradise Dove wrote:
I'm using an A101u and it seems to work fine connected to a
Carrier Access Access Bank I (24 FXS).
How did you get it working with asterisk?
The directions provided by Sangoma are very clear:
- compile and install libpri and zaptel
-
This guy found out what I was planning to do as a boxed solution.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of asterisk
h323
Sent: Thursday, December 09, 2004 2:32 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Get rid of H323 problems for 100$
On Thu, 2004-12-09 at 14:23 +0200, Shoval Tomer wrote:
Hi.
Please excuse me asking this again. But it really puzzles me.
Asking multiple times does not change a proper answer.
We're installing asterisk at a branch office at NJ (HQ is at
Petach-Tikva)
It'll need to support 5 POTS lines, 11
I've been running an Asterisk box with 4 FXO ports and 12 FXS ports for
months. The cards are sharing interrupts. The machine has one network
card too. The system behaves very well. In my experience, putting
multiple TDM cards in one box works. I've not been so lucky with
multiple T1/E1 cards,
Adam Goryachev wrote:
Hi all,
I have the opportunity to demo asterisk to a large group of people, and
was just wandering *how* to do that?
ie, I can put a couple of phones on a desk, which looks nice, but
doesn't really look exciting, because they are just phones so, how
do you 'demo' the true
Since we haven't heard much since the alpha six weeks ago, here's a
reminder:
http://voip-info.org/wiki-Asterisk+bounty+bluetooth+cell-phone+support
This bounty is now up to $500 for full functionaly, and $300-$400 for
partial.
___
Asterisk-Users
Froogle found me one supplier for the SPA-841, not sure I trust them
though. Does this phone even exist yet? Does anyone have any
experience with it? Does anyone know a vendor other than
Atacomm/voipsupply?
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[EMAIL
Any thoughts/suggestions would be greatly appreciated.
Adds, moves, and changes which are the bane of any telephone administrator.
Show how fast it is to add an extension with voicemail. Using AMP, I can add
a new SNOM in under a minute with voicemail. Contrast that with the 15-20
minutes or so
In an environment with multiple asterisk boxes, each with a 4PRI card
and 4PRIs (92 Zap ports) and oversubscription on SIP peers, is there a
way to get the asterisk box to use the Zap interfaces on another box in
times of congestion? While the oversubscription ration would be
optimized for the
Im using the adit 600 with a cmg card and 5 8 port FXS cards connected to a
MetaSwitch VP3510 via ethernet. just plugged the cmg card into the same
ethernet lan as the softswitch. Signaling is mgcp (even supports g729 for
the first 24 calls - can do all 40 with a different cmg card). here's
On 08-Dec-2004, Alex Barnes wrote:
The reason its probably not working is because your Xlite is sending the
request to the Asterisk.
The Asterisk isn't a SIP proxy hence all it does is see if it recognises
the addressee.
This isn't strictly true. A SIP proxy is one solution to this demand,
On Thursday 09 December 2004 14:22, Eric wrote:
Hi Sean,
Thanks for your reply, but that wasn't exactly what I was getting at.
I don't need to increase the system's imposed limit on the number of
open files. I'm more concerned to see if anyone has run across a
memory or fd leak in asterisk
Jay Milk wrote:
Froogle found me one supplier for the SPA-841, not sure I trust them
though. Does this phone even exist yet? Does anyone have any
experience with it? Does anyone know a vendor other than
Atacomm/voipsupply?
Jay,
I just talked to someone at Voxilla who told me the phone should
I am hoping to replace my Nortel 8x24 switch with
Asterisk. Right now my cabling comes from my outside
phone box into my office and into a punchdown block
and leaves the punchdown block as an amphenol
connector which currently plugs into the Nortel
swicth. A second amphenol connector them plugs
I don't need to increase the system's imposed limit on the number of
open files. I'm more concerned to see if anyone has run across a
memory or fd leak in asterisk that sucks them all up.
My apologies. If you are looking for leaking fd's in asterisk, I am
afraid I am not much help.
I never could get attended transfer to work with the BT-100 on 1.0.5.16. Where
did you get 1.0.5.18? It's not anywhere obvious on Grandstream's web site.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: Thursday, December 09, 2004
we are about to deploy an asterisk server. on the external side we will
have an ISDN30e plugged in to a E100P card. On the internal side i wish
to use a channel bank. Which products work best for this solution? Can
another E100P be used? and if so... what channel banks are compatible?
where can
On Thu, 2004-12-09 at 08:19 -0700, Damon Estep wrote:
In an environment with multiple asterisk boxes, each with a 4PRI card
and 4PRIs (92 Zap ports) and oversubscription on SIP peers, is there a
way to get the asterisk box to use the Zap interfaces on another box in
times of congestion? While
I am hoping to replace my Nortel 8x24 switch with
Asterisk. Right now my cabling comes from my outside
phone box into my office and into a punchdown block
and leaves the punchdown block as an amphenol
connector which currently plugs into the Nortel
swicth. A second amphenol connector them
On Thu, 9 Dec 2004, Leif Madsen wrote:
One problem I can think of in regards to the pin is that each
participant would need their own unique pin number if that is what you
are going to associate their sound clip with in the database. This
leads to more and more pins being used as you add
On December 9, 2004 10:30 am, Richard Reina wrote:
I am hoping to replace my Nortel 8x24 switch with
Asterisk. Right now my cabling comes from my outside
phone box into my office and into a punchdown block
and leaves the punchdown block as an amphenol
connector which currently plugs into the
On Thu, 9 Dec 2004, Dan Goscomb wrote:
we are about to deploy an asterisk server. on the external side we will
have an ISDN30e plugged in to a E100P card. On the internal side i wish
to use a channel bank. Which products work best for this solution? Can
another E100P be used? and if so...
Hey folks,
Using FreeBSD 5.2.1 and I've got the current zaptel driver installed
from ports (0.8_1) and current ports asterisk (1.0.1). I've set
options HZ=1000 in my kernel config, recompiled and rebooted and as far
as I can tell, I've done everything right but when I try to use the
On December 9, 2004 10:38 am, Dan Goscomb wrote:
we are about to deploy an asterisk server. on the external side we will
have an ISDN30e plugged in to a E100P card. On the internal side i wish
to use a channel bank. Which products work best for this solution? Can
another E100P be used? and if
Hi,
What is out there in terms of SIP enabled handsfree speakerphones?
Looking for something that works well and also fits a low budget.
I am used to using a Cisco 7940. It is a great phone but a bit expensive.
Thought about the Polycom SoundPoint 300 until I realized that it does not
include
I bought one, I was going to get two but they were charging £50
delivery *each* box!
which is rather extortionate! So I only went for one in the end.
I'll buy a 2nd processor from somwehere else at some stage I think.
Mike
On Thu, 9 Dec 2004 14:53:50 -, Asterisk [EMAIL PROTECTED] wrote:
In an environment with multiple asterisk boxes, each with a 4PRI
card
and 4PRIs (92 Zap ports) and oversubscription on SIP peers, is there
a
way to get the asterisk box to use the Zap interfaces on another box
in
times of congestion? While the oversubscription ration would be
optimized
On 09/12/2004 05:54 Steven Critchfield said the following:
Unless Symbian has branched off of cell phones, I doubt it. SIP on a
cell phone right now doesn't make sense.
the nokia 9500 (communicator) as sold in asia uses symbian and has built-in
802.11b. i can see a software SIP phone here being
I have installed successfully more then four cards in a machine before.
I had a firewall with eight network interfaces (one quad card, one duo
and two singles)
I have machines with two dialogic boards, a pci display card, and a
network interface.
And I know I've had machines at home that had
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