On Wednesday 28 May 2003 15:30, Gary wrote:
Has anyone consider or found an opensource ILS ??
doubt it.
from http://www.collaborium.org/onsite/peru/christian/h323/tsld027.htm:
- ILS is a Microsoft extension to the LDAP protocol
- It is used to make it easier for Netmeeting users to find each
Gary wrote:
Has anyone consider or found an opensource ILS ??
just thinking of same and possible link with asterisk
.
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IIRC there was a
how you mean "board is not full duplex"? ofcourse is full duplex it has
been used for iptelephony "long" time ago.
have you try this board in linux? you have drivers?
i have two cards ;)
Thomas
Benjamin Miller wrote:
This board is not full duplex and will not work with *. I have 3
Title: Message
"Full
Duplex" means that it has enough voice resources to send and receive data at the
same time.
It has
been used in telephony for a long time but usually as a call recorder or an IVR
application that requires only half duplex.
I have
installed the GlobalCall driver and
What network card are you using ? (model and vendor)
Martin
On Tue, 27 May 2003, Nick Eggleston wrote:
We are using the zaptel driver to deliver a combined voice/data T1 circuit.
The data channel-group is using the cisco hdlc protocol (on the linux side)
and connects with a cisco router on
Digium T100P
On Wed, 28 May 2003, Martin Pycko wrote:
What network card are you using ? (model and vendor)
Martin
On Tue, 27 May 2003, Nick Eggleston wrote:
We are using the zaptel driver to deliver a combined voice/data T1 circuit.
The data channel-group is using the cisco hdlc
1) you need two accounts in iconnecthere
2) you need to register with two accounts
3) then simply receive the call using one and send it over another account
Martin
On Wed, 28 May 2003, pradeep kumar wrote:
Hi All.
I am trying to setup asterisk so that I can place two outbound calls via
If you don't have any hardware for conferencing than you could use the
ztdummy from zaptel package. Check the archives. look for ztdummy
Martin
On Tue, 27 May 2003, Rahul Gupta wrote:
Hello ,
I am a newbie to * and have just been able to call
a sip User Agent on a different machine thru
When immediate is set on a port that is an fxo, what is the meaning of this ?
Will it go immediately to the s extension of the context when the line
first rings, or something else ?
I am looking for a way to stop the line ringing two extra times before
being answered by the channel bank.
Has anyone consider or found an opensource ILS ??
just thinking of same and possible link with asterisk
IIRC there was a shell-backend to OpenLDAP that would allow to use slapd
as an ILS-Server. Ask Google for details. I think it was called
Netmeeting Directory Kit.
There is information on
FXO ports don't get DID numbers usually so they'll always go to 's'
Martin
On Wed, 28 May 2003, Jon Pounder wrote:
When immediate is set on a port that is an fxo, what is the meaning of this ?
Will it go immediately to the s extension of the context when the line
first rings, or something
1. Voicemail, and the voicemail itself will be stored on another box, NFS
mounted, or I might use mysql. There will be a little bit of call routing
via iax to a separate * box with a channel bank on it.
2. I don't disagree with you, they do throw in a lot, but redhat does have
its advantages,
Cool, dual CPU it is =) I'll watch out for the frame buffers.
Other than that, are there any other known issues with doing this? I will
also be using dual-channel memory...
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer
Sent:
Check the archives, someone got the openline card working. I had been
talking with voicetronix about writing a driver for the openswitch cards
but they haven't been interested recently.
dave
On Wed, 28 May 2003, Daniel ANDRE wrote:
Hello,
I would like to know if voicetronix card
You may be able to solve this with either multiple mount points or
symlinks to alternate disk partitions. Or, a really ugly method
would be to write an quickie AGI that parses the output of du -s
/var/spool/asterisk/voicemail/blah and if that number is higher than
the permitted value, new
Nope.
usecallerid=no should work for it.
If not you might try to modify the code in chan_zap.c
Martin
On Wed, 28 May 2003, Jon Pounder wrote:
my question was - will immediate put an end to the extra 2 rings before
pickup ?
(I know they go to s eventually.)
At 10:40 AM 5/28/2003 -0500,
Sorry, resolved it myself. Because I need to load
all modules manually I did not load enough codecs and formats.
- Original Message -
From:
Jan Boon
To: [EMAIL PROTECTED]
Sent: Wednesday, May 28, 2003 5:46
PM
Subject: [Asterisk-Users] Newbie:
--- Gary [EMAIL PROTECTED] wrote:
I just had a thought, but haven't tried it yet
we are able to include one conf file into another...
this leads to the thought can we use the same file
to include into
iax.conf as well as sif.conf ?
does the various differences actually have an
On Wed, 2003-05-28 at 11:02, Joe Antkowiak wrote:
1. Voicemail, and the voicemail itself will be stored on another box, NFS
mounted, or I might use mysql. There will be a little bit of call routing
via iax to a separate * box with a channel bank on it.
2. I don't disagree with you, they
I would like to know if maxsilence and silencethreshold parameters in
voicemail.conf work only for voicemail2 application and what are the main
differences that exist between voicemail and voicemail2. What possible
values silencethreshold can take?
Thanks
On Wed, 2003-05-28 at 11:02, Joe Antkowiak wrote:
1. Voicemail, and the voicemail itself will be stored on another box, NFS
mounted, or I might use mysql. There will be a little bit of call routing
via iax to a separate * box with a channel bank on it.
Don't use Mysql. if you ever have had
On Wed, 2003-05-28 at 13:30, Steven Critchfield wrote:
Don't use Mysql. if you ever have had to deal with it in a production
environment that works it over, you will know that as it reaches it's
limits, it starts a death spiral that is very difficult to recover from.
For our software on a
On Wed, 2003-05-28 at 14:30, Steven Critchfield wrote:
Don't use Mysql. if you ever have had to deal with it in a production
environment that works it over, you will know that as it reaches it's
limits, it starts a death spiral that is very difficult to recover from.
For our software on a
I've run tests with MySQL. It is very fast unless you have
many processes that need to access it at once and then
the weak point is it's locking method which let's only
one process in at a time So you run into a wall as the
size/complexity of your project grows past some point.
But for a PBX
On Wed, 2003-05-28 at 14:11, Ryan Butler wrote:
On Wed, 2003-05-28 at 13:30, Steven Critchfield wrote:
Don't use Mysql. if you ever have had to deal with it in a production
environment that works it over, you will know that as it reaches it's
limits, it starts a death spiral that is very
On Wed, 2003-05-28 at 14:36, Ron Gage wrote:
On Wednesday 28 May 2003 02:30 pm, Steven Critchfield wrote:
On Wed, 2003-05-28 at 11:02, Joe Antkowiak wrote:
1. Voicemail, and the voicemail itself will be stored on another box,
NFS mounted, or I might use mysql. There will be a little bit
I'll save my typing fingers somewhat on this one - you are doing great
arguing about all the crappiness of mysql and actually backing it up with
real examples. It is nice to see that for a change in comparison to all the
mysql lovers that love it just because but have no basis to compare it to
Hello
I'm starting to learn about Asterisk and trying to install the first one.
I've a doubt. Here in Brazil the Telecom Standards uses E1. So, I'll use a
Wildcard E100P to inter-connect with 30 POTS.
Should I use a Channel Bank or simply a Multiplexer?
Whats the difference between a
Make sure that you don't have a R2 signalling. Since then you'll have
problems EuroISDN PRI is all right.
Martin
On Wed, 28 May 2003, Ricardo Saar Gemignani wrote:
Hello
I'm starting to learn about Asterisk and trying to install the first one.
I've a doubt. Here in Brazil the
On Wed, 2003-05-28 at 15:09, Jon Pounder wrote:
I'll save my typing fingers somewhat on this one - you are doing great
arguing about all the crappiness of mysql and actually backing it up with
real examples. It is nice to see that for a change in comparison to all the
mysql lovers that love
Sorry, what is R2 signalling?
When I've that?
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, May 28, 2003 5:43 PM
Subject: Re: [Asterisk-Users] About Channel Banks
Make sure that you don't have a R2 signalling. Since then you'll have
On Wed, 2003-05-28 at 15:34, Ricardo Saar Gemignani wrote:
Hello
I'm starting to learn about Asterisk and trying to install the first one.
I've a doubt. Here in Brazil the Telecom Standards uses E1. So, I'll use a
Wildcard E100P to inter-connect with 30 POTS.
Should I use a Channel
has anyone ever used an 827-4v DSL router to do SIPtoPOTS conversion? how would I set
up my 4 pots lines to be SIP extensions/phones? any ideas? Cisco site is a little
lacking on sample configs. I would like to set up 3 of the ports for FXS analog sets
and one port for FXO. convert all the
Don't use Mysql. if you ever have had to deal with it in a production
environment that works it over, you will know that as it reaches it's
limits, it starts a death spiral that is very difficult to recover from.
For our software on a dual P3 866 with a gig of ram, the limit was
around 1.5
My example of heavy load where mysql could not even begin to handle the
situation was a project with real time stock market data streamed in as
bids and offers and trades happened, statistics computed from that in real
time, database kept in sync live, and charts and graphs plotted in real
was that mysql 3.23.x or 4.0.x ?
michael
*** REPLY SEPARATOR ***
On 28.05.2003 at 16:18 [EMAIL PROTECTED] wrote:
My example of heavy load where mysql could not even begin to handle the
situation was a project with real time stock market data streamed in as
bids and offers
Steven Critchfield wrote:
Don't use Mysql. if you ever have had to deal with it in a production
environment that works it over, you will know that as it reaches it's
limits, it starts a death spiral that is very difficult to recover from.
For our software on a dual P3 866 with a gig of ram, the
was that mysql 3.23.x or 4.0.x ?
michael
I did most of my mysql work some time ago with 3.x. I have, however,
installed mysql 4.0.12 since I'm working on a CASE tool which needs to
support both mysql and postgresql. I'll be the first to admit that
mysql has probably improved (a lot?) since
On 28 May 2003, Steven Critchfield waxed:
8's
While I'm on the postgres bandwagon for now, I wouldn't want it in the
middle of a phone system doing heavy call loads either. Postgres also
has some downsides too. As I understand it, postgres doesn't understand
prepared statements, or at least
I think what Benjamin is trying to say is that the Dialogic D/4 is not
capable of full duplex operation when reading/writing from the PCI bus.
It's capable of full-duplex operations only when using it's own SC-BUS.
Yes, you're right these cards have been used for VOIP. But, to achieve
that
We've got an asterisk system hooked up to a number of telephones via a channel
bank.
[*]T100P---CAC(access bank)---Phones...
What we are seeing is problems with DTMF tone detection on some (but not all) of
the FXS ports. In our testing, we plugged a simple analog phone into each of
the first 10
Hi. I need to send calls to different programs depending on where the
call originates. For example, I need calls from San Diego (NPA 619 and
858) to to be routed differently than L.A. calls. I tried entries like:
exten = 4044633/_619.,1,OurApp,sandiego-queue
exten =
I don't think you can use wildcards in the ANI matching areas, though
I'd be happy if this were the case.
You'll probably need to write an AGI that hands back an appropriate
variable set to something that a Goto can parse.
The use of wildcards in ANI matches would be darn handy, though.
JT
On Thu, 22 May 2003, Robert Hajime Lanning wrote:
Shared IRQs are just bad. Just the PC hardware does not have enough IRQs
to handle all the devices we want to connect to them, nowadays.
No, that's not the reason at all. Most systems are not short of IRQs.
The reason is that standard PCI bus
I was just thinking that. Shouldn't this be a feature?
I'm sure coding it would be a cut and past job. :)
Another one for the TO-DO list Mark.
:)
Jamie
On Wed, 28 May 2003 16:45:44 -0700
John Todd [EMAIL PROTECTED] wrote:
*This message was transferred with a trial version of
This build is about 200 years old. I've tried it and it
actually doesn't work for me. CVS builds work fine
though.
Jamie
On Wed, 28 May 2003 00:40:44 +1000
Shaun Ewing [EMAIL PROTECTED] wrote:
*This message was transferred with a trial version of
CommuniGate(tm) Pro*
- Original
On Fri, 23 May 2003, Stephen R. Besch wrote:
own card to determine if it is the interrupting hardware.
In point of fact, this same strategy was possible on the ISA bus, it
just wasn't used.
I dont think level-triggered interrupts ever reliably worked on ISA, it
was always edge. I never heard
I have just a second case where after * had been up for a few weeks working
perfectly, it suddenly stopped talking to all the extensions. The I4L was
still working perfectly though...
I tried to stop asterisk, rmmod the modules, and insmod, and re-start
asterisk but it did this:
Using
Title: SIP INVITE and ACK go to different ports
Greetings,
CVS 05/23/03. 10.50.4.140 is an * box. I see SIP INVITE to port 5060 and ACK (after OK) to port 32824.
The log is attached.
tcpdump shows
18:31:48.380006 10.50.4.140.5060 wmssqa02.webley.5060: udp 615 (DF)
18:31:48.390007
On 24 May 2003, Thilo Salmon wrote:
The other issue is a legal one. In order to connect to the incumbent
telco your equipment has to be certified. I believe unless quite a few
of us get together, this one might be a real problem.
Easy solution -- Have * talk to SS7-certified equipment. Cisco
From Reading http://www.cisco.com/univercd/cc/td/doc/pcat/827.htm that
device is H323 and only has FXS ports. So any of the Cisco Docs on H323
would probably point you in the right direction.
Shawn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dave
Then why bring it up?
What has any of this to do with the original topic?
Jamie
On 27 May 2003 12:35:57 -0500
Steven Critchfield [EMAIL PROTECTED] wrote:
*This message was transferred with a trial version of
CommuniGate(tm) Pro*
I had a similar problem when there was timing problems
with my
No! You're outnumbered and trapped!
On Wed, 28 May 2003 10:22:33 -0700 (PDT)
Patrick Tabor [EMAIL PROTECTED] wrote:
*This message was transferred with a trial version of
CommuniGate(tm) Pro*
--- Gary [EMAIL PROTECTED] wrote:
I just had a thought, but haven't tried it yet
we are able to
I've got an end-user in Australia that needs some gear .. I can ship it
over there, but it'd be a bit of a pain to find adapters localized to AU,
even though I know stuff is all 110/220 these days.
Anyone know a good cheap place to buy a Cisco ATA186 and maybe a
Netgear/etc dsl router/switch
exten = 4044633/_619.,1,OurApp,sandiego-queue
exten = 4044633/_858.,1,OurApp,sandiego-queue
exten = 4044633/_213.,1,OurApp,losangeles-queue
exten = 4044633/_.,1,OurApp,default-queue
but it didn't seem to work. Every call went to the default queue.
Take out the _. rule and just leave it
No, that's not the reason at all. Most systems are not short of IRQs.
The reason is that standard PCI bus only has 4 interrupt lines.
Shared IRQs are not bad at all. It's only bad if the card is badly
designed.
All the Zaptel hardware *should* be able to share IRQ's. Each of the
cards has
Be sure you do *not* have NAT mode turned on.
Mark
On Wed, 28 May 2003, Alex Zarubin wrote:
Greetings,
CVS 05/23/03. 10.50.4.140 is an * box. I see SIP INVITE to port 5060 and ACK
(after OK) to port 32824.
The log is attached.
tcpdump shows
18:31:48.380006 10.50.4.140.5060
Mark Spencer wrote:how is this passed to the script (I know 0 perl)
I would have though something like ./vmail.cgi?context=mycontext
but that makes the script barf
try logging in as [EMAIL PROTECTED]
Same results for me:
1. I can see things in my INBOX, which is in the default context,
yeah logging in like that works, however the stored mail seems to be in the
default context not the one the mailboxes are really in.
eg: if I move my mailbox entry to default, I see my mail when I login, if I
leave in the context it was always in, I see no mail after I login.
is this some sort
Shouldn't the priority also be different for each entry?
This would make it:
exten = 4044633/_619.,1,OurApp,sandiego-queue
exten = 4044633/_858.,2,OurApp,sandiego-queue
exten = 4044633/_213.,3,OurApp,losangeles-queue
exten = 4044633,4,OurApp,default-queue
This should work I would think. Give it
On Wed, 28 May 2003, Mark Spencer wrote:
No, that's not the reason at all. Most systems are not short of IRQs.
The reason is that standard PCI bus only has 4 interrupt lines.
Shared IRQs are not bad at all. It's only bad if the card is badly
designed.
All the Zaptel hardware *should* be
On Thu, 2003-05-29 at 02:36, [EMAIL PROTECTED] wrote:
On 24 May 2003, Thilo Salmon wrote:
The other issue is a legal one. In order to connect to the incumbent
telco your equipment has to be certified. I believe unless quite a few
of us get together, this one might be a real problem.
The
Hello All,
I am using an E100P card on a PRI
line. I need to setup a FAX extension. Can somebody help me please?
Marco
That's about right, inlcuding postage.
- Original Message -
From: nathan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 12:44 PM
Subject: [Asterisk-Users] What is the going rate for the Snom 100 in the UK?
Hi All,
What is the going rate for the Snom 100 in the
Very weird problem,
got a channelized T1 with SBC. There is a toll free number that points
to a local number on that T, when the local number is called, *
recognizes dtmf tones, when the toll free number is called, * does not
pick up dtmf..
any thoughts?
lizardbox
Nathan,
Get in touch with www.provu.co.uk ask to speak to Tim, and tell him you heard from me
(Andy Powell) that they had a deal running where you could get Snom 100's for 140
gbp...
HTH
Andy
*** REPLY SEPARATOR ***
On 29/05/2003 at 12:44 nathan wrote:
Hi All,
What is the
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