I have that on my HandyTone 286, and it's got some issues. At least
mine does. The page loading in FireFox very rarely ever completes, and
this means that I can't really provision this thing very well at home
(I only run Linux). I brought it to work and it loads up fine in IE
here.
I engaged
On Wed, 16 Mar 2005 19:58:55 -0800, Luki [EMAIL PROTECTED] wrote:
Anyway, if anyone ever needs this info, they can Google it now :-).
Might be a good thing for the wiki too. ;)
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If you have any FXS ports, use wcfxs.
On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi
[EMAIL PROTECTED] wrote:
Hi,
I was using a TDM400P with cvs version of asterisk, loading the driver
with modprobe wctdm.
Some days ago I switched to stable version 1.0.6, where I found no
trace of
Can you run dmesg after that command and tell us what the relevant output is?
On Fri, 18 Mar 2005 15:32:02 +0100, Alessio Focardi
[EMAIL PROTECTED] wrote:
Hello Dana,
Friday, March 18, 2005, 3:23:36 PM, you wrote:
DO If you have any FXS ports, use wcfxs.
No, only green modules.
But
Click on Reply in your mail client, type message, click Send.
On Fri, 18 Mar 2005 16:15:24 -, Kanishka Somaratne
[EMAIL PROTECTED] wrote:
Hi
how do i reply a question asked in this mailling list.
tks
Kanishka
___
Asterisk-Users mailing
I tried using the iLBC codec, and whlie I like it, I ran into a
strange issue. I did a few searches on Google and haven't found anyone
with the same issue as this.
Anyhow, I was using a Plantronics analog headset and box plugged into
a Digium TDM card, dialed out over my VoIP provider's IAX
On Mon, 21 Mar 2005 08:27:17 -0500, Steve Clark [EMAIL PROTECTED] wrote:
snip
It doesn't sound very professional.
comedian
n 1: a professional performer
snip
What's not professional about that? :)
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On Mon, 21 Mar 2005 10:01:04 -0800, Sys Admin [EMAIL PROTECTED] wrote:
I am setting up a new asterisk based call center. I just read:
http://www.voip-info.org/wiki-IAX+versus+SIP
After reading this and other google results for IAX vs SIP is there
any reason why i should use SIP anywhere !!
If your IP address changes daily, you need to continually update this, correct?
I too have this issue with SIP, but my ISP is a little unstable, so I
need to use DynDNS or No-IP to kinda work around/with it. I assume you
can't specify a hostname there?
--
Dana
On Mon, 21 Mar 2005 13:45:44
On Tue, 22 Mar 2005 12:10:17 +0400, Jean-Michel Hiver
[EMAIL PROTECTED] wrote:
On a LAN where NAT is not an issue I would go for SIP + decent
hardphones with good echo cancellation.
On the internet with all sort of NATs + Firewalls, IAX is a must but
unfortunately I don't know of any good,
Heh. It was really funny. If I hadn't seen the preview of the email in
Gmail, I probably would've missed this thread entirely. Thanks to you
all and Gmail preview for making my day!
--
D
On Mon, 21 Mar 2005 18:40:42 -0800, Luki [EMAIL PROTECTED] wrote:
Nice, gUys...
Don't pick on Raphael too
On Mon, 21 Mar 2005 21:36:53 -0800, Geoff Nordli [EMAIL PROTECTED] wrote:
Hi Everyone.
Asterisk is one of those applications that need to be built from cvs on a
regular basis to keep up with the changes. I have always used package
management tools like apt.
How does everyone manage their
Yeah, if we're making a list, add DNS name resolution to that list. :)
--
Dana
On Tue, 22 Mar 2005 23:48:46 +0400, Jean-Michel Hiver
[EMAIL PROTECTED] wrote:
If the IAXy had a bit more work done on it, it could be a good option,
but it's not at the current time.
Yep! Things like:
-
On Tue, 22 Mar 2005 12:49:06 -0500, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
I'm very, very confused. Dialing out, through VoicePulse, with both gsm
and ulaw CODECs, most of my calls are great. However, calling my
(non-Asterisk) voicemail at my job, and calling my cell phone both
produce
this issue :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dana Olson
Sent: Montag, 21. März 2005 17:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] iLBC codec and mute issues
I tried using
On Wed, 23 Mar 2005 06:54:03 -0500, Time Bandit [EMAIL PROTECTED] wrote:
6) Configuration requires Linux, as opposed to a web browser or
something more standard.
I compiled iaxprov on Cygwin, works nicely. There's somebody on this
list that made a Windows version to provision it. Works
On Wed, 23 Mar 2005 13:59:07 -0500, Time Bandit [EMAIL PROTECTED] wrote:
There's another feature request. Let me dial ### or something to find
my IP...
That's not something to do with the IAXy, you can make an AGI script
that will tell you your IP. I had this script somewhere but I can't
On Wed, 23 Mar 2005 12:55:45 -0800, Sys Admin [EMAIL PROTECTED] wrote:
so seems like the verdict is go IAXy with a IAX only network ? Most of
the problems of the IAXy device seems like will be fixed with firmware
updates and wont require a hardware update..
this way we get the advantage of a
On Wed, 23 Mar 2005 16:13:12 -0500, Time Bandit [EMAIL PROTECTED] wrote:
So how am I going to provision the device in the first place, to be
able to dial this extension, if I don't even know the IP?
Oups, sorry, didn't think about this one.
Check winiaxyprov, the version 1.01 can scan your
On Thu, 24 Mar 2005 08:09:19 -0700, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Dana Olson wrote:
On Wed, 23 Mar 2005 06:54:03 -0500, Time Bandit [EMAIL PROTECTED] wrote:
6) Configuration requires Linux, as opposed to a web browser or
something more standard.
I compiled iaxprov
On Thu, 24 Mar 2005 15:34:26 -, Giles Coochey
[EMAIL PROTECTED] wrote:
How about scanning for it's mac address?
http://ipscan.sf.net/ipscan.exe
--
http://www.umich2.com
Digium doesn't label the MAC address on the device, unless it's such a
fine print that no one
On Thu, 24 Mar 2005 09:34:48 -0600, Art Zemon [EMAIL PROTECTED] wrote:
Folks,
Please forgive my ignorance. I think that what I am asking must be so
obvious that no one bothers to write it down. But I don't know the
answer so...
I want to set up * with one incoming VOIP phone number. If
On Thu, 24 Mar 2005 12:45:58 +0100, Fred Blaise [EMAIL PROTECTED] wrote:
Hello all
I have come to Asterisk with no previous telco experience.
As I will be playing with Asterisk really soon, I would like to have
some pointers as to some tutorials in telco that could help me get into
all
On Thu, 24 Mar 2005 13:03:46 -0700, JD Austin [EMAIL PROTECTED] wrote:
xlite doesn't seem to have this problem.
X-Lite doesn't support IAX.
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On Sat, 26 Mar 2005 04:14:54 +0200, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Mar 23, 2005 at 04:37:02PM -0500, Dana Olson wrote:
My company has thousands of entries in the DHCP server, and it would
take forever to go through each and every one of them. Not to mention
that I, being
On Thu, 31 Mar 2005 10:00:12 -0500, David Brodbeck
[EMAIL PROTECTED] wrote:
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
My understanding is that to an extent when we buy Sangoma
we're putting the dagger to Digium.
If anything puts the dagger to Digium
On Thu, 31 Mar 2005 10:04:34 -0500, Kanuri, Seshu (Company IT)
[EMAIL PROTECTED] wrote:
Folks!
I want to let everyone know that I have been trying to migrate from
1.0.6 to 1.0.7 last few days and I have come across serious issues in
the build 1.0.7. What I found are listed below. I would
I've been meaning to try it again. A large number of builds have been
sent since I last tried. And boy, it was sooo slow and more
resource-intensive than its Windows counterpart.
I haven't been using a softphone at home because I'm waiting for
GnomeMeeting w/SIP to get into Ubuntu or Debian.
Do you think Digium has enough money for that? I don't know how large
Sangoma is, but they've been around almost as long as I have...
I think it'd be a great idea though, if they have the cash for it.. I
bet it'd pay off too.
On Thu, 31 Mar 2005 11:00:48 -0500, mattf [EMAIL PROTECTED] wrote:
On Thu, 31 Mar 2005 11:37:19 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
My understanding is that to an extent when we buy Sangoma
we're putting the dagger to Digium.
If anything puts the dagger to Digium it'll be their own inability to
engineer reliable hardware.
I
Well, I'm certainly not selling xten.. Perhaps my enthusiasm extends
from my disgust with everything else. In particular, kphone, and
sjphone. I have noticed latency with xten in meetme, but if I just dial
somebody it works better than anything I've tried (so far.. I've only
spend
On Thu, 31 Mar 2005 10:01:29 -0600, Henry Devito [EMAIL PROTECTED] wrote:
An additional fault in 1.0.7 When you log into voicemail and select advanced
options there are none. On previous versions it would ask if you would like
to send a message, etc.
That's strange man. My VM works fine on
On Thu, 31 Mar 2005 14:49:11 -0300, Carlos Gabriel Drach
[EMAIL PROTECTED] wrote:
Kris Edwards wrote:
This is the best linux sip phone I've used so far. Audio quality has
been perfect and it seems really stable, so hopefully it will be out of
beta soon.
I might actually pay for the full
On Mar 31, 2005 8:55 PM, Bruno Hertz [EMAIL PROTECTED] wrote:
Brian Capouch [EMAIL PROTECTED] writes:
Hmmm. I just got the latest beta build, which identifies itself as 1105d.
The keypad functionality is perfect.
Hmmm. Good for you. We were talking about sjphone, though :)
Regards,
On Mar 31, 2005 1:24 PM, Dana Olson [EMAIL PROTECTED] wrote:
On Thu, 31 Mar 2005 10:04:34 -0500, Kanuri, Seshu (Company IT)
[EMAIL PROTECTED] wrote:
Folks!
I want to let everyone know that I have been trying to migrate from
1.0.6 to 1.0.7 last few days and I have come across serious
On Mar 31, 2005 1:44 PM, Dana Olson [EMAIL PROTECTED] wrote:
On Thu, 31 Mar 2005 11:37:19 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
My understanding is that to an extent when we buy Sangoma
we're putting the dagger to Digium.
If anything puts the dagger to Digium it'll
On Apr 7, 2005 6:20 AM, Matteo Brancaleoni [EMAIL PROTECTED] wrote:
Sangoma doesn't do that. they don't sell directly, thus allowing
resellers to have a money gain and pay the time to support the end
user.
Actually, they do sell directly. I emailed them a short time ago, and
they gave me
On Apr 11, 2005 8:11 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
nat=no
disallow=all
allow=g729
allow=g726
auth=plain
context=default
canreinvite=yes
username=USERNAME
secret=PASSWORD
dtmfmode=info
fromdomain=REALM
fromuser=USERNAME
qualify=1000
insecure=very
I am using
On Apr 12, 2005 9:38 AM, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
Actually I guess what I am looking for is semi-sealed box that I can add
1 or 2 PCI cards too. A regular PC work work in most cases since I do
not want a keyboard or mouse attached to it. I do not want users
screwing with the
On Thu, 20 Jan 2005 15:26:15 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
While it will probably be handled when you move out of outlook, please
wrap your lines at a reasonable length.
Please tell me that Gmail is fine... If it isn't, I'll have to find
something else.
As of right now,
You can use pretty much any headset you want. I use just a regular,
inexpensive Labtec one from WalMart for now. Works fine.
As for softphones... I just tried out SJphone yesterday and I like it more.
I only tested it out briefly on my work Windows XP system, and haven't tried
it at home on Linux
I can't tell you why it's failing, as I don't know. But to answer your other
question, I have firmware 1.0.5.22 that I found from a link a short time ago
on the mailing list.
I'm having some issues with DHCP and the BudgeTone phone, as it doesn't seem
to like the TFTP options we put in. (I do
1.0.5.
In my searches, I've found that it's possibly due to not having a
sound card in the system. Do I require a sound card, or is there an
alternate way around?
Thanks in advance.
--
Dana Olson
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High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp.
Is that right?
--
Dana
On Fri, 28 Jan 2005 14:06:11 -0600, Eric Wieling aka ManxPower
[EMAIL PROTECTED] wrote:
Dana Olson wrote:
I've been reading
On Fri, 28 Jan 2005 14:12:26 -0600, Eric Wieling aka ManxPower
[EMAIL PROTECTED] wrote:
Dana Olson wrote:
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp.
Yes. Something else must be wrong, like
-- voip-info.org
On Mon, 31 Jan 2005 20:21:51 -, Richard Dutton [EMAIL PROTECTED] wrote:
Hi Guys,
I know no doubt this has been covered on the list a zillion time before, but
can anyone point me to some good resources on using Asterix as a VoIP
gateway?
I would like to get two
I would be interested in this. I have Delphi 6 I believe. I'm a little
rusty, but I'd like to make a really basic client. If you have any
pointers along with that DLL, that would be fantastic.
--
Dana
On Tue, 1 Feb 2005 08:28:39 -0500, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
That's fine if
On Thu, 3 Feb 2005 10:44:18 -0800, Gareth J. Greenaway
[EMAIL PROTECTED] wrote:
I am encountering a problem with the voicemail portion of asterisk, when
someone goes to forward a voicemail and they choose the option to prepend the
message with a greeting, the call never ends, eventually this
I'm looking at ordering a server from HP. I checked around on Google
and found in the Wiki that the ProLiant DL380 is supposed to be known
to work with *.
I'm going to get a price quote on the following setup:
HP ProLiant DL380 G4 Server w/ the following options:
Intel Xeon 3.20GHz/1MB
2GB REG
on
the forum about problems with the DL380 G4 the TE410P.
See this thread:
http://lists.digium.com/pipermail/asterisk-users/2005-January/081544.htm
l
Cheers,
Edge.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson
Sent: Friday
On Fri, 4 Feb 2005 19:07:14 +0100 (CET), Christoph Rothe
[EMAIL PROTECTED] wrote:
On Fri, 4 Feb 2005, Dana Olson wrote:
I'm looking at ordering a server from HP. I checked around on Google
and found in the Wiki that the ProLiant DL380 is supposed to be known
to work with *.
HP
On Fri, 4 Feb 2005 14:35:14 -0500, Dana Olson [EMAIL PROTECTED] wrote:
On Fri, 4 Feb 2005 19:07:14 +0100 (CET), Christoph Rothe
[EMAIL PROTECTED] wrote:
On Fri, 4 Feb 2005, Dana Olson wrote:
I'm looking at ordering a server from HP. I checked around on Google
and found in the Wiki
On Fri, 04 Feb 2005 15:53:13 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Fri, 2005-02-04 at 16:02 -0500, Dana Olson wrote:
Out of curiosity, do any of you think that the ML350 would be a better
choice, with similar options? They're a lot cheaper too, and I haven't
found any
On Mon, 07 Feb 2005 11:08:19 -0800 (PST), Job 317 [EMAIL PROTECTED] wrote:
I have asterisk installed on a linux workstation (1 phone in and 1 phone
out jack).
I want to configure Asterisk to show me any available data about any
calls (i.e. phone numbers, caller-id) as well as screen unwanted
On Tue, 8 Feb 2005 11:56:18 +0200, Yousri Farouk [EMAIL PROTECTED] wrote:
Hello all,
i would like to configure TDM11B with Asterisk, if any one have the
configuration steps please provide me it.
Thanks in advance
Have you tried looking at Digium's site??
On Mon, 7 Feb 2005 23:16:05 -0600, Eric Rees [EMAIL PROTECTED] wrote:
Has anyone seen this message trying to install an TDM400.. spurious
8259A interrupt: IRQ7
This error happens after I do a modprobe wctdm and then the system
hangs. I am installing this in an Asus motherboard with a VIA
I run Debian, and it's not hard to get a base install running. If you
want a GUI and such, then it'll be more than follow the screen
prompts. I've been writing some Debian documents, if you're
interested, email me off-list.
Anyhow, on pretty much any distro, you can make your own packages
(RPM,
On Tue, 8 Feb 2005 12:30:36 -0500, Noah Miller [EMAIL PROTECTED] wrote:
I installed a tdm400p into a old p2 machine.
I'm not able to see it under /proc/interupts or using lspci..
we removed all other cards. changed slots, forced irq to that slot..
etc etc.
what is the min specs needed
On Wed, 9 Feb 2005 08:23:57 -0800 (PST), Paul Chan
[EMAIL PROTECTED] wrote:
Hi All,
I just installed Asterisk 1.0.5, and the
installation went fine (I ran modprobe zaptel and
modprobe wcfxo). However, when I ran ztcfg I get the
following:
ioctl(ZT_LOADZONE) failed: Invalid argument
On Wed, 9 Feb 2005 11:44:30 -0500, Paul Rodan [EMAIL PROTECTED] wrote:
I know FWD uses a SER/Asterisk combo, and I keep hearing about the massive
benefits, however, my initial playing around in SER's configuration
indicates it's NOTHING like Asterisk at all, and almost 5x as difficult to
Yeah, I wouldn't mind having a look at it as well.
--
Dana
On Wed, 9 Feb 2005 09:36:27 -0800, Michael Levenson
[EMAIL PROTECTED] wrote:
Why not share with the community?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Chandler
Sent: Wednesday,
On Wed, 9 Feb 2005 13:46:41 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
Guys, Im new to asterisk and voip but Im have a couple of questions
regarding the initial setup.
1. Im going to install an asterisk server at home, where I have 2 phone
lines, what kind of card do I need to get? I was
On Wed, 9 Feb 2005 14:59:44 -0500, Noah Miller [EMAIL PROTECTED] wrote:
Hi Walid -
I need to use Asterisk to call out PSTN numbers via an analogue line. I
understand Digium manufactures these kinds of cards, but can someone
tell me
which model number it is. I really only need a card
And for Windows, a minimum purchase of 1 unit... Is he using Mac or
PocketPC? If not, then he doesn't have to worry.
On Wed, 09 Feb 2005 15:59:45 -0500, Sergey Kuznetsov
[EMAIL PROTECTED] wrote:
Adrian,
Have you ever read the note for that?
=head Comment
* NOTE
G.729a
On Thu, 10 Feb 2005 11:44:37 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Thu, 2005-02-10 at 11:36 -0500, Noah Miller wrote:
IMO, your best defence is leaving ssh's default setting
which disallows root logins entirely. There's no reason
for a remote user to ever have to log in
On Thu, 10 Feb 2005 17:49:23 +, Clive Carter
[EMAIL PROTECTED] wrote:
I hesitated before sending this, as I have been flamed before for being a
beginner. but
I am newish to linux/asterisk, and I am running an ssh server. It is still
running with default settings, (I dont know yet
On Thu, 10 Feb 2005 14:05:38 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
Derek Whitten wrote:
I also call bullshit.. OpenBSD does NOT allow ssh root login by
default.. why do you think that they have such an excellent security
track record..
Derek,
I am sorry to say,
On Fri, 11 Feb 2005 09:00:49 -0500, Dana Olson [EMAIL PROTECTED] wrote:
On Thu, 10 Feb 2005 14:05:38 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
Derek Whitten wrote:
I also call bullshit.. OpenBSD does NOT allow ssh root login by
default.. why do you think that they have
Could anyone shed any light on how SER and/or Asterisk (stable branch)
has held up for them in that last while?
Are you using SER and/or * in a production environment? Do you ever
restart the software or reboot the system? How many users are
utilizing the system? How many calls per
. The only issue we have is that the audio
(greeting or message) being play from Asterisk sometimes has a
robotic or stuttering quality to it. I suspect this is latency in the
data network but I have yet to figure it out.
-Steve
Dana Olson wrote:
Could anyone shed any light on how SER
On Mon, 14 Feb 2005 20:01:18 +0100, Bruno Hertz [EMAIL PROTECTED] wrote:
Another point to note is that seemingly all closed source softphones
(SJ, XLite beta and also cornfed) make connections to web servers
and transmit platform/call information. Don't know how you think about
that, but for
There are tons of comments about the S100i, or IAXy, as it's called.
Can't say I like it myself, we have 3 of them right now, and they get
really hot and for some reason don't have the MAC address labeled on
them, and also we couldn't get them to actually take an IP from a
Microsoft DHCP server
I've had the best luck with SJphone. Using a USB headset really helps
as well, opposed to the sound card or onboard audio with a standard
headset. I don't like the skins that come with most softphones, and
XLite is no exception. SJphone lets me disable it, and the profiles
are nice, since you can
Have you made any progress on this?
--
Dana
On Mon, 24 Jan 2005 11:38:52 -0500, Jon Bebeau [EMAIL PROTECTED] wrote:
I'm exactly in the middle of benchmarking the A104 and T410p. I'm
developing a matrix of CPU, bandwidth throughput and trying to find high
water marks under several loads;
I'm looking to purchase and deploy a bunch of hardphones for agent
use. The phones will have to register with Asterisk and/or SER,
depending on where the phones go. They need only one line, G729 codec,
and no super fancy features. Preferrably something that is easy to
provision.
I would think the
I've called using G729 SIP phones over my LAN, and I think it sounds
quite good. YMMV.
On Fri, 04 Mar 2005 15:58:23 -0500, Martin Roy [EMAIL PROTECTED] wrote:
I have 2 Asterisk servers connected with IAX. It's working fine I can
call an extension from one phone in an office to another phone in
Thanks for your replies.
My main concern is to keep the price down. If the BudgeTones are crap
phones, which previous posts to this mailing list seem to indicate,
and we have to replace them often, then the price for a better phone
would be worth it. I don't think we need 3-way calling either,
On 4/19/05, ruben cuevas rumin [EMAIL PROTECTED] wrote:
I would like to know if for this simple test (communication using IP
address directly) , need I a dialplan or no??? And if I need a
dialplan, where I could obtain any example of a extension.conf file
for this simple test. (because I only
Hello. I checked in the wiki and read a bunch of old threads from this
mailing list but haven't found what I'm looking for.
I'm using a simple PHP script, and here is the relevant portion:
fputs($socket, Action: Monitor\r\n);
fputs($socket, Channel: Zap/1-1\r\n\r\n);
That works fine. As does
Alright, thanks you guys. I was hoping to not have to do that, but I
guess it's time to get my PHP on. I find myself re-learning it every
time I start a new project. I love the language, I'm just forgetful.
:-)
___
Asterisk-Users mailing list
Well, I guess that I'm not as good as I once was... If anyone would
care to assist me in this, I would appreciate it. If you want to
contact me even on IM, IRC or off-list, anything's cool with me... I
would really appreciate the help.
--
Dana
___
I'm using Debian Stable both at home and at work, and Asterisk runs fine for me.
Use what you know.
--
Dana
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To UNSUBSCRIBE or update
For agent queues, I checked this option out:
; leastrecent - ring interface which was least recently called by this queue
When I use this, if I have 3 agents logged into the queue, and I pump
in 3 calls to the queue, I would expect that one call would go to each
agent, but instead, it doesn't
I found that if I dynamically add, for example SIP/8000, to a queue,
then calls in the queue will sorta pile up on the 9 extensions on that
phone - not what we want to happen.
If I log in to the queue using AgentLogin, then the behaviour is as
expected - one call at a time.
Is there a way around
Did you try dialing out over ZAP/g1?
On 4/22/05, Mark Phillips [EMAIL PROTECTED] wrote:
I have a full PRI installed on my * machine. I can get inbound calls
just fine but can't make outbound ones.
Zaptel.conf says;
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
zapata.conf says
I've had an issue with my 286 ever since I got it. Basically, the web
interface doesn't load, and I can't make any calls - although I get
dialtone. Also, I can call it and it will ring. But I get no audio.
The main issue is that I can't get into the web interface anymore... I
did once, but not
On 4/26/05, Sean A. Newton [EMAIL PROTECTED] wrote:
On Mon, 25 Apr 2005, Wiley Siler wrote:
Call waiting can be disabled in Asterisk via *71 regardless of the phone
used.
Cheers,
Wiley
Well, this is part of a larger problem I'm having.
I can't get CheckGroup/SetGroup to work as
Another option is the AudioCodes Mediant 2000 devices. I have one with
8 T1/E1 ports, one with 4 ports, and one with 1 port. You can also get
up to 16 ports, and there is a 2-port model as well. I am using SIP
with them, but I think you can use H.323 with a different firmware.
I am using them in
On 4/27/05, Sean A. Newton [EMAIL PROTECTED] wrote:
On Tue, 26 Apr 2005, Dana Olson wrote:
You mean like the problem I described earlier on this list?
http://lists.digium.com/pipermail/asterisk-users/2005-April/103153.html
I am not sure why I didn't think of disabling call waiting
On 4/27/05, Guy Boehm [EMAIL PROTECTED] wrote:
Hello,
I want to call a peer over the Asterisk Manager with this php-script:
html
body
PRE
?
$socket = fsockopen(192.168.204.44,5038, $errno, $errstr,
$timeout);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName:
On 4/27/05, Paul Shiflet [EMAIL PROTECTED] wrote:
I just received my TDM400 card from digium with 2 fxo and 2 fxs
interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS
phones. How do i interface my POTS phones with this; can i just crimp an
RJ45 connection on the end of the phone
You would need a transcoding license between the Asterisk PBX and the
G711 phone...
On 4/28/05, Matt [EMAIL PROTECTED] wrote:
So
[g729 provider] -(SIP or IAX)--- [g729 asterisk server]
This is how I'd be setup.. actually more like this:
[g729 provider] --(sip)
On 4/29/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi !
I am using queues with MOnitor Application but the thing is that Iwant to
save the files starting with the Answering agent name. I have tried a lot
of things but nothing seems to work. If i put Monitor application on top
of dialing
60 calls a day is nothing. I'm sure your Asterisk box can handle it
with the standard Monitor command.
I've recorded many calls, 8+ hours straight and I'm on a crap old
Pentium 3 633MHz system.
What exactly do you fear will happen if you record on the Asterisk box?
--
Dana
On 4/29/05, Steve
Wouldn't introducing Samba into the mix be even worse?
I would think it would add more processing power and network use to be
constantly writing over the network as opposed to recording on the
same box.
If it's such a critical system, it should have the power to do that,
but that's not the
On 5/3/05, Matt Riddell [EMAIL PROTECTED] wrote:
Chris Mason wrote:
Why not?
Because you have not licensed the file for broadcasting across your
telephone network.
Don't jump to conclusions. ;)
How many other people are there here that write music? Would there be
any interest in
I'm trying to figure out a way to make my own agent login, because I
don't like how the default works.
I have the login and logout working fine using the dynamic add and
remove commands, but I need to be able to create a list of users and
passwords.
I thought of a way to do it using a list of
On 6/29/05, Joe Murray [EMAIL PROTECTED] wrote:
Is anyone on this list using and audiocodes FXO gateway? I have
Asterisk(1.07 on OS X) setup and working fine, including SIP phones
and IAX2 phones - I can make outbound calls just fine and receive
inbound calls just fine. However, I can't seem
I think they were hoping that the client would connect to Asterisk,
which makes it kinda useless, really.. But connecting Asterisk to the
Gizmo network is handy.
--
Dana
On 7/4/05, Adrian A [EMAIL PROTECTED] wrote:
I have a Gizmo account working perfectly in my Xten Eyebeam, so there
should
I installed a vanilla 2.4.31 kernel from kernel.org and my system was
working great.
Then I tried upgrading zaptel to 1.0.9 and now I get unresolved symbols:
# modprobe zaptel
/lib/modules/2.4.31/misc/zaptel.o: /lib/modules/2.4.31/misc/zaptel.o:
unresolved symbol proc_mkdir_R8712438a
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