day...
Regards,
Adam
- --
Adam Goryachev
Website Managers
www.websitemanagers.com.au
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.6 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFI3JsIGyoxogrTyiURAi2UAKCGuoNdby+4hSipuVnfaBi6onXfdQCgquSV
Yp4eDzhjNg48M
will
receive the fax, and then send it, instead of just transparently acting
as a gateway ?
perhaps, in between the receive and send you might use scp to copy the
file to the remote asterisk server, and then ssh to create the call file
on the remote asterisk
Regards,
Adam
- --
Adam Goryachev
different (assuming you really are
testing your system properly).
Just my 0.02c worth.
Regards,
Adam
--
Adam Goryachev
Website Managers
Ph: +61 2 8304 [EMAIL PROTECTED]
Fax: +61 2 8304 0001www.websitemanagers.com.au
Lenz wrote:
[queuetransfer]
exten = _0.,1,DigitTimeout(5)
exten = _0.,2,ResponseTimeout(10)
exten = _0.,3,Answer
exten = _0.,4,NoOp
exten = _0.,5,NoOp
exten = _0.,6,SetCallerPres(prohib)
exten = _0.,7,Dial(Zap/g1/${EXTEN:1})
exten = _0.,108,NoOp(Got busy here)
As you can see, a transfer from a
Shawn Kelley wrote:
Hi,
Does anyone know how to do a re-map of a key on the Polycom to make it
dial a number.
I know how to remap a key to a certain function, but I don’t know how
to make it dial a number.
I’m wanting to re-map the “Service” key to dial *8 for a group pickup.
Any help
On Thu, 2006-02-23 at 02:13 -0600, Anton Krall wrote:
Now thas confusing to me.. How do you actually take 16 calls at a time? I
see 301's have 2 line keys.. And each can handle 16 calls... How do you
actually take all 16 and switch between all of them?
Hmmm, top-posting...
anyway, pretty slow
On Wed, 2006-01-25 at 14:10 -0600, Kevin P. Fleming wrote:
Aaron Daniel wrote:
We had the bug on 1.2.2, but when I rolled back to 1.2.1 to fix the
problem, everything started working. Doesn't seem like it's a bug in
1.2.1 :)
It is not. The bug was introduced during the 1.2.1-1.2.2
On Mon, 2006-01-23 at 23:00 -0700, Douglas Garstang wrote:
Polycom SoundPoint 601 has 4 'lines'. :)
Actually, it has 6 'lines' :)
Needing a 4 line phone is going to decrease your choices of phones.
Why do you need 4 lines?
He probably hasn't worked out the difference
On Mon, 2006-01-23 at 20:46 -0500, Jeff Herring wrote:
Issue: horrible echo (and squeals, and underwater-like sound) on speaker
phone when calling from extension to extension.
Is it a direct call from one extension to another, or a meetme, or
something similar?
echo not present when calling
On Mon, 2006-01-23 at 16:26 -0500, Bill Gibbs wrote:
I know the Polycoms work with NAT, but you have to specify the public
IP.
No you don't, at least, I never have, and it works perfectly for me
every time I have a client who regularly moves their polycom 501
from home - work and back
On Mon, 2006-01-23 at 15:34 -0500, Franklin Webb wrote:
Basically I have phone representatives that log into one of several
queues (not using chan Agent, we log in by the extension), and
frequently these agents have to make attended transfer calls to
outside numbers. This transfer basically
On Mon, 2006-01-23 at 12:16 -0500, Steve Totaro wrote:
Is this also true for recording of calls? Will I require licensing for
each recorded call? Will the server see a big performance hit in this
setup whether or not a license is required?
In my experience (which was using asterisk 1.0.x at
On Fri, 2006-01-20 at 21:20 -0500, Michael Miller wrote:
I have over 50 Asterisk servers geographically distributed in pairs all
connected via DUNDi. Contact me off list and I will be happy to describe
my experience.
Would love to hear about peoples experiences like this.
Also, what are the
On Thu, 2006-01-19 at 16:16 +1100, James Harper wrote:
My alternatives then are:
1. a modem to dial up the internet and send email to an email to sms
gateway
2. a subscription to a dialup sms service (Telstra do offer one)
3. a GSM modem
4. SMS to fixed line
#3 means no reliance on fixed
On Sun, 2006-01-15 at 10:16 +1100, James Harper wrote:
If they won't, you can basically do the same thing by dialing out from
asterisk on one pstn line coming back in through a second pstn line,
and
using the asterisk milliwatt generator. Or, if you have another
asterisk
system
On Mon, 2006-01-09 at 11:40 -0600, Rich Adamson wrote:
It would be very interesting to know the real numbers that have it working.
The archives (and about two/three years of attempting to help others with
the exact same problem) suggests no better then maybe one in ten or twenty
will ever
On Tue, 2006-01-03 at 12:42 -0600, Brent Torrenga wrote:
I use IP Kall to forward my missed cell phone calls to. This way, if my
phone is off, or out of a service area, calls will go to my * box.
Concurrently, all incoming calls to my * box cause it to dial my local
extensions at home, my
On Mon, 2006-01-02 at 09:35 -0800, Don Fanning wrote:
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises
Silva
Action: Originate
Channel: SIP/13 -- this should be the first phone you want to ring
(your own phone usually)
I don't want it to ring a REGISTERED device
On Fri, 2005-12-30 at 20:04 +0530, [EMAIL PROTECTED] wrote:
Thanks a lot Mr. Alexander Lopez for your prompt attension.
I tried the same thing but it wouldnot happen. I use it as:-
exten = 12,1,Dial(Agent/12)
exten = 12,2,Hangup
where agent 12 is configured as :-
agent = 12,12, vivek
On Wed, 2005-12-28 at 14:00 -0300, Javier Ergas wrote:
I believe this behavior has nothing to do with the [EMAIL PROTECTED] Scripts.
I think the
problem is in the PRI signalization.
I can see the zap hangup messages when trying to call a disconnected number.
.
-- Executing
On Thu, 2005-12-29 at 16:22 -0300, Javier Ergas wrote:
I have tried both inband and outofband too unsuccessfully. I think the
priindication parameter says how Asterisk reports Busy and Congestion
to the PSTN, not the other way around.
In the Asterisk config sirrix.conf
On Mon, 2005-12-19 at 07:21 -0600, Kevin P. Fleming wrote:
Matthew wrote:
For the uninitiated among us (myself included) what is ACD login/logout
support?
The Polycom phones can send XML NOTIFY messages to signal to the server
the agent is logged in/out/paused. I know of no
On Thu, 2005-12-15 at 09:33 -0800, John Biundo wrote:
I'm particularly worried about acceptance of this shared line (or lack
thereof) aspect of the system. My wife will get the idea of
extensions, transfers, parking, etc. because she uses a PBX at work,
though I worry that the habits of
On Sun, 2005-12-11 at 16:18 +0200, Warren Burstein wrote:
I'm running asterisk 1.0.9 with TDM400B's for both internal and external
lines.
I put in the macro that dials outside lines an AbsoluteTimeout(36000),
never expecting it to happen. But it does, a few times a month.
I've noticed
On Tue, 2005-12-06 at 21:41 -0600, Jerry Jones wrote:
Just in the process of figuring this out myself. i do have it working
on an IP601 with a sidecar. Here are my notes.
On the polycom
Create a contact directory entry for the extension you wish to
monitor. Yes the contact must match
Would like to find out if it is possible to setup a VoIP network with
asterisk and 9 x polycom IP 501 or 600 handsets, the main difficulty is
that there is no cabling in place, and it isn't possible to run cabling
(heritage building, need to demolish walls + half the ceiling to get the
cables in).
On Fri, 2005-11-25 at 13:08 -0500, Gary MacKay wrote:
After playing around with the CALLERID(number) and
CALLERID(name) variables and things, I find that asterisk is sending
the name to my phone and the name is unknown. I added a line
exten = _X.,Set(CALLERID(name)=${CALLERIDNUM}) and now it
On Sat, 2005-11-26 at 10:43 -1000, Jean-Denis Girard wrote:
Hi list,
I installed iaxmodem and Hylafax to see how it compares to rx/txfax; so
far I had 0 failure in my limited testing with a Philips HFC21 fax
machine that failed very often with txfax (same test platform, with
On Wed, 2005-11-23 at 08:30 -0700, [EMAIL PROTECTED] wrote:
Does anyone know of a brute force that will work on a serial interface like
hyperterminal?
Look at expect... you should be able to throw something simple together
using a shell + expect script...
ie, connect and
expect login:
send
On Wed, 2005-11-23 at 12:53 -0800, Anthony Rodgers wrote:
Hi Dave,
exten = callpark,1,Dial(SIP/1000) didn't work - invalid extension
What about:
exten = callpark,1,Dial(Local/[EMAIL PROTECTED])
Regards,
Adam
___
--Bandwidth and Colocation
On Thu, 2005-11-24 at 10:26 +, Julian Lyndon-Smith wrote:
I posted a similar problem a couple of days ago, and one of the
responses suggested that the TE4xxP may be on it's way out.
Is there any way of testing this card to see if that may be the case ?
Speak to digium and ask them how to
What firmware version did you use for the polycom phone ??
I just tried it on my IP600, and when I press the park button, it waits
for me to dial an extension number, then I press park again, and it just
hangs up the call.
Thanks,
Adam
On Tue, 2005-11-22 at 13:56 -0800, Anthony Rodgers wrote:
On Sat, 2005-11-19 at 00:07 -0600, [EMAIL PROTECTED] wrote:
Hi. I'm a new user of Asterisk. My question is:
I want to log outbound calls in a database ( postgres ). Everything is OK
except that asterisk always marks calls to my FXO iface ( Zap/4 ) as
answered as soon as it accepts to dial the
On Sat, 2005-11-19 at 18:02 +0100, Michael Kenjie Nukui wrote:
Hi, i have this Wildacard FXO in my [EMAIL PROTECTED] box, connected to
POTS. When i make an incoming call, it takes about 8 to 10 rings
before my card pick up the incoming call and answers it. Here is my
config. Can somebody
On Tue, 2005-11-15 at 23:11 +, Julian Lyndon-Smith wrote:
Done. Done. Done :)
http://bugs.digium.com/view.php?id=5762
Julian.
In case anyone is interested, this has worked like this for at least 18
months, I had the same problem, so I just changed to using Monitor
before dropping the
On Tue, 2005-11-15 at 09:35 -0800, trixter aka Bret McDanel wrote:
On Tue, 2005-11-15 at 11:56 -0500, Jason Pyeron wrote:
a unicode document comes in two flavors UTF8 and UCS2 in windows UTF8
may work, but UCS2 cannot work, as it is 2 bytes per character.
UTF8 will not work if wordpad
On Wed, 2005-11-16 at 11:26 +, Julian Lyndon-Smith wrote:
Bugger. I had hoped that it was a recent bug ..
As a matter of interest, how do you know which agents have answered the
call ? I liked having the agent number as part of the monitored filename.
Julian.
Well, in my case, there
On Thu, 2005-11-10 at 14:44 -0700, Kyle Hagan wrote:
Here is the actual messages file:
Nov 9 15:19:12 xeonAsterisk kernel: BUG: soft lockup detected on CPU#3!
Nov 9 15:19:12 xeonAsterisk kernel:
Nov 9 15:19:12 xeonAsterisk kernel: Modules linked in: md5 ipv6
parport_pc lp parport
On Thu, 2005-11-10 at 11:40 -0600, Doug wrote:
At 11:27 11/10/2005, Jason Brashear, wrote:
Can this be done?
I have a customer service que that if full go to v-mail.
I would like to know how I can put two e-mail address for it to go to.
Is that possible?
My suggestion for this is
On Thu, 2005-11-03 at 13:08 -0500, BJ Weschke wrote:
Well, I hope many people feel that way about it. :-)
The best thing to do at this point is to download and test the betas
of 1.2 right now so we can get 1.2 released and we can move on to fun
things like app_followme post 1.2.
Any
On Tue, 2005-11-01 at 14:35 +0800, Steve Underwood wrote:
That patch may or may not work. It just depends on which day you grabbed
the asterisk code. This is why I stopped reponding to requests about the
makefile patch failing. It is simply impractical to offer a working
solution for
On Tue, 2005-10-18 at 14:35 +0200, [EMAIL PROTECTED] wrote:
Hi,
This issue has been discussed probably a million times on every asterisk
forum in the world and I have the same problem too. Another problem you would
have with the agents is that when they make an outgoing call they are not
However, some channels on one of the channel banks are still problematic.
I'm checking with Rhino to see if it's a channel bank problem, since
the noise always appears on the same channel no matter how many times I
reboot, unload/load etc.
It has been said that a power-off + power-on is
On Mon, 2005-10-03 at 17:54 -0400, Matt Roth wrote:
List members,
2) What will happen on the NFS client if the NFS server crashes (I expect the
leg files to be written to the local mount point until the mount is
reesablished)?
Why don't you create a file on the NFS server called something
On Fri, 2005-09-30 at 10:51 -0500, Kevin P. Fleming wrote:
Matt wrote:
A post-install would be great (or I myself can write a script)... it
isn't that big of a deal.. I just wanted to see if I was over looking
something. Tagging the sound directory for a version would also be
good
nicely for both, the IP300
is watching 601, but isn't working
Has anyone got a IP300 phone to display the status ?? Any suggestions
for things to look at/etc ??
PS, of course, the current state is that 600 is off-hook and all others
are on-hook.
Regards,
Adam
--
--
Adam Goryachev
Website
On Sun, 2005-09-04 at 07:39 -0500, Derrick Stensrud wrote:
Re-sending your message every 12 hours isn't nice wait at least a
couple of days, and while you wait, try to read/test more things, so
that the second time around, you can actually demonstrate that you have
progressed somewhat
On Sun, 2005-09-04 at 14:09 -0700, Adrian A wrote:
Hi all,
I'm trying to setup a simple IVR menu in a context in extensions.conf.
So far, I have:
extension s for playing back the menu
# to repeat it
* for directory
0 for operator
1 which goes to another context: exten =
On Mon, 2005-09-05 at 01:31 -0400, Kurth Bemis wrote:
I am attempting to assemble a proposal for a client of mine that is
looking to replace their phone system. I think it's a good first
installation with 4 POTS incoming and 15 extensions, with an overhead
paging system. I also think that
On Wed, 2005-08-24 at 15:04 +1000, Michael Felder wrote:
Hello Craig,
Yes I would like to dial 0 to get an outside line and dial tone, then
dial the number.
I have Polycom IP600 and IP 500s.
Mike
Just wondering how people who use 0 to access an outside line deal with
the following
On Tue, 2005-08-16 at 23:53 -0700, Pudenz, Duane wrote:
We are testing our Asterisk server prior to deployment. The server has
a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and
one PRI for local calls.
We are using sipp from two different stations routing a test number
On Sat, 2005-08-13 at 19:53 -0400, Jeff Buchbinder wrote:
Hi; I've been using Asterisk for a few months now, and I have run into
an interesting issue that I thought someone else in the community may
have run into:
I have an Asterisk install set up to receive helpdesk calls, route
them to
somewhere?
Regards,
Adam
--
--
Adam Goryachev
Website Managers
Ph: +61 2 8304 [EMAIL PROTECTED]
Fax: +61 2 8304 0001www.websitemanagers.com.au
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Asterisk-Users
Jonathan k. Creasy wrote:
YeahI think that every install I have done the first thing that
happens is why is there a delay before the call connects? and the
answer is you have to hit dial or wait 10 seconds.
What all phones does that apply to? I'm fairly certain it applies to the
On Wed, 2005-08-03 at 09:47 -0400, John Novack wrote:
Tim Litwiller wrote:
I've been using * at home at my house for while and like it but for
work I didn't know the answers to these questions.
But now my new employer is wanting to upgrade a very old phone system
and wants to make
On Mon, 2005-07-25 at 07:43 +0100, Julian Lyndon-Smith wrote:
Many thanks, Niklas.
I'll use this as a basis and let you know how things pan out.
To get a better spread out load (across your various apps) look at the
random application, you should be able to get each call to randomly go
to
On Mon, 2005-07-25 at 23:15 +1000, Michael Felder wrote:
Hello,
I have configured my polycom ip600 and ip500.
The phone works well.
But the clock is wrong and flashes the whole time. Drives me nuts!
I have set the time offset on the DHCP / boot server. 36000 (I'm in
Australia!)
It
On Sat, 2005-07-23 at 06:35 -0400, Joseph wrote:
exten = _6XXX,2,Busy
exten = _6XXX,3,Hangup
But the whole point is that I don't want the caller to hear a busy
signal or get hung up, I want the Queue to try the next available agent.
Which it does at the moment, just with the errors
On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote:
Adam Goryachev [EMAIL PROTECTED] wrote:
On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote:
For this reason, I believe that if a fork were
ever necessary, it would struggle to beat a distinct path away from
the Asterisk Binary
On Sat, 2005-07-23 at 12:00 +0300, Tzafrir Cohen wrote:
Disclaimers aside, who has the copyrights in those cases?
Do you actually read the emails on this list? or just like to jump right
in and help the brawl continue? The disclaimers don't affect copyright,
the author of the work/patch/source
I have a queue setup using Asterisk CVS and roundrobin, however calls
seem to be distributed in the same way as rrmemory (round robin with
memory), ie, it is alternating between the two people in the queue
rather than always calling the first available person in the queue
first.
I am using agents
On Wed, 2005-07-20 at 10:39 -0700, Victor Rini wrote:
David Stude wrote:
#2, I'm planning to interface Asterisk with a Norstar MICS via PRI. Can
anyone recommend a reference book or site more suited to this task?
Sorry that link is kind of dead.
I have the pdf if anyone is
On Thu, 2005-07-21 at 09:59 +0200, Alessio Focardi wrote:
PF Oh, you mean the completely natural feeling put them on hold, dial
PF new party, tell them you have a transfer, hit transfer? I want some of
PF whatever kool-aid the person who thought that one up had. I still feel
PF like I'm
On Thu, 2005-07-21 at 09:22 -0400, Waldo Rubinstein wrote:
On Jul 21, 2005, at 9:04 AM, Eivind Trondsen wrote:
1) send sound to the caller of an ongoing call
2) retain control so the call can be terminated based on a timer (or
whatever)
Any tips would be greatly appreciated! Thanks in
On Thu, 2005-07-21 at 15:56 -0400, Adam Dobrin wrote:
I'm using Polycom 501's; with stable1.0.8, g729 and a very decent
machine; we have a PRI interface to a T1.
Many users complain that after a given amount of time, say, 30 or 40
minutes on a call, the outside party complains that their
On Thu, 2005-07-21 at 13:14 -0600, Colin Anderson wrote:
From Slashdot
http://slashdot.org/articles/05/07/21/0135213.shtml?tid=126tid=95 :
One of the points made is that there is sometimes no way to tell the
location of a VOIP phone, which is a problem if you are unable to talk.
How about
On Thu, 2005-07-21 at 15:30 +0100, Asterisk wrote:
I've got several agents on a queue. However, they often forget to go
not ready or log off when they can't answer the phone.
I would like a person calling my queue to be on the queue for a max of 2
minutes, and I'm using the rrmemory
On Thu, 2005-07-21 at 18:32 -0700, Lee Howard wrote:
Kevin P. Fleming wrote:
You seem to be neglecting the amount of work that Digium puts into the
Asterisk (and related) products on an ongoing basis that is given to
the community at no charge.
So at least we agree, then, on what the
On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote:
It has been flippantly said, a number of times, that if you don't
like the situation then you can fork the project. A major fork seems
(to me) to be pointless for one main reason (and a couple of lesser
reasons):
As I see it, anyone
On Thu, 2005-07-21 at 21:36 -0700, Nguyen Trung Tin wrote:
Hello ALl
i need context to do:
record to wave file and receive DTMF when recording wave file.
for example:
exten = s,1,Record(test:wav)
exten = s,2,hangup
when recording, press # to hangup and i want to receive others DTMF
(while
On Wed, 2005-07-20 at 14:42 +0300, Tzafrir Cohen wrote:
On Wed, Jul 20, 2005 at 07:01:48PM +0800, Craig Guy wrote:
How do you handle:
RTF
Not very common
Isn't this easily converted to text?? I thought the format for this was
pretty simple, but I could be wrong...
Disclaimers
beeps, and
then the phone hangs up. I ask him if the base station is plugged in,
and I then hear something along the lines of Oh... ummm, yeah...
thanks, cya...
Regards,
Adam
--
--
Adam Goryachev
Website Managers
Ph: +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396
the manager API to terminate the call if their credit reaches zero,
connect and process active channels on an regular basis (as needed), use
the AGI to reduce the credit by the needed amount at the end of the call
(from h extension, or g option to Dial).
Regards,
Adam
--
--
Adam Goryachev
Website
On Wed, 2005-07-20 at 13:47 -0500, Matthew Boehm wrote:
Per my conversation below with digium, are there any legal alternatives
to digium's G729? It is out of date, and doesn't support VAD nor silence
detection.
Well, I guess it supports what it is supposed to, ie, g729a :) since the
On Tue, 2005-07-19 at 17:05 +0100, Bob Goddard wrote:
On Tuesday 19 Jul 2005 14:45, Martin Sutherland wrote:
Silly question, you did restart * when you put the .so in the correct
directory (normally /usr/lib/asterisk/modules) and it has the correct
permissions?
Does show g729 respond with
On Mon, 2005-07-18 at 09:25 -0600, Aaron with Morad wrote:
I have been searching for a while and can't find anything specific
like this.
Here's is my setup:
IAXy -- broadband network -- Asterisk -- TE110P -- Channel
Bank -- POTS lines (FXO)
Everything works fine except for
On Tue, 2005-07-19 at 00:35 +, Obelix wrote:
I have been reading a number of the past threads about G.729 licensing., about
how the registration keys are linked to the network configurations, limited
number of registrations etc, etc.
Is there no reason why the decoding can't be done in
On Tue, 2005-07-19 at 04:50 +0100, Kevin Walsh wrote:
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
I dunno... people seem all up in arms about this but honestly I fail to
see the problem. Digium is doing what they can to make money and provide
services while keeping Asterisk as free and
On Mon, 2005-07-18 at 23:04 -0500, Kristian Kielhofner wrote:
The new firmware and bootrom already require 4mb flash, which the 301,
501, and 600 have. You can't load firmware 1.5.2 on the 300 or 500!
Are you sure of that?? I don't recall seeing that noted anywhere... and
I'm sure I've
On Tue, 2005-07-19 at 14:20 +1000, Michael Felder wrote:
Is the 301 and 501 available in Australia?
Yes, I ordered and received some already.
Regards,
Adam
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I'm am writing a small php program to pull some info out of our
Asterisk's queue_log. I'm having trouble figuring out what some of
the parameters mean.
Here's an example:
1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||Ray Balbin 25
(716)250-3405
1st field is current
as a single call. That would then
show that it really is quite reliable ie, I could have just been
lucky for that 2 minutes that I left the connection up for
Feel free to ask me questions on my config/setup, or to run further
tests...
Regards,
Adam
--
--
Adam Goryachev
Website Managers
Ph
for tftp and ftp with newer bootrom
now... but I would still prefer FTP ...
Also, apparently they also support https, which I would prefer even
more, but I haven't tried it as yet... (I think this only works on the
301/501 and 600 as well)...
Regards,
Adam
--
--
Adam Goryachev
Website Managers
Ph
are looking for...
Regards,
Adam
--
--
Adam Goryachev
Website Managers
Ph: +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au
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Adam Goryachev
Website Managers
Ph: +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au
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http
If I'm doing a time include in extensions.conf, do I want 04:00-23:00
and 23:00-04:00 or 04:00-22:59 amd 23:00-03:59? I want to make sure
that at no time are both or neither included.
In other words, does the second time go to HH:MM:00 or HH:MM:59?
At one time when I was hassling people
I just needed to test a dialup modem connection (don't ask) and I had a
modem connected to a TDM card (FXS port) which then dialled out via a E1
PRI on a TE4xxp card.
See my log below:
atdt0198xx
CONNECT 36000 V42bis
** Dial IP **
Username:
Password:
Entering PPP Session.
IP
. Of course, perhaps someone
should check this, as we can't transfer a call until after we accept
it...
Regards,
Adam
--
--
Adam Goryachev
Website Managers
Ph: +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au
[31521]
Username = heath
Secret = heath
Type = friend
Qualify = 600
Defaultip = 172.28.184.105
Context = sip
Nat = no
AFAIK, the username should match the [] at the top, eg:
[heath]
Username = heath
Secret = heath
Type = friend
Qualify = 600
Defaultip = 172.28.184.105
Context =
), if that is
the case, then you need to look at QoS.
Regards,
Adam
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in the asterisk extensions.conf?
what in the phones phone.cfg
what in the phones sip.cfg ?
I would love to be able to support/use this.
Thanks for any info you can provide.
Regards,
Adam
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On Thu, 2005-06-23 at 14:41 -0500, Brian West wrote:
chan_woomera is another alternative h323 implementation.
visit www.pbxfreeware.org for more information.
Without being rude, why do we need another one? ie, why did you decide
that another one needed to be written, what are the advantages
-written (re-invented
if you like) or else they really aren't important to anyone anyway
Just my 0.02c worth
PS, why would you need to host it on sourceforge anyway, why not just
stick it on your own website ??
Regards,
Adam
Regards,
Adam
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Website Managers
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On Wed, 2005-06-22 at 11:46 -0400, Adam Robins wrote:
I installed a TDM400P with 4 FXO modules. Before moving all of my
office phone lines to it, I decided to move only one for testing. I
plugged it into port 4 on the card.
When I launch an outbound call as ZAP/g1/${EXTEN}, Asterisk goes
On Wed, 2005-06-22 at 17:49 -0400, Mike M wrote:
On Wed, Jun 22, 2005 at 05:19:47PM -0300, Alessandro wrote:
But all ports are green!
Really? Maybe they aren't making the RED FXO cards anymore. You should
look at them carefully for p/n differences and not rely on colors. The
zapel
On Wed, 2005-06-22 at 21:59 -0700, George Pajari wrote:
Adam Robins asked:
Shouldn't [Asterisk] be smart enough to go to Zap/4 as the only available
port in the group [with a live trunk]?
Adam Goryachev wrote:
No, asterisk doesn't do dialtone detection.
But this isn't an issue
On Tue, 2005-06-07 at 23:28 -0300, Joshua Colp wrote:
A network booboo occurred and and just like it warns (note the word
WARNING), it received a mini frame before the first full voice frame...
Nothing too serious, audio might sound odd for less then a second but it
should recover.
Actually,
On Mon, 2005-06-06 at 07:17 -0400, Andrew Kohlsmith wrote:
On Friday 03 June 2005 05:50, Chris Coulthurst wrote:
I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used
line. Would the following work for 911 calls?
Why would you do this? Use a group:
Yes, use a group...
On Fri, 2005-06-03 at 08:28 -0600, Rich Adamson wrote:
I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used
line. Would the following work for 911 calls?
[e911]
exten = 911,1,ChanIsAvail(Zap/1)
exten = 911,2,Dial(Zap/1/911)
exten = 911,3,Hangup()
exten =
On Fri, 2005-06-03 at 13:50 -0700, Chris Coulthurst wrote:
Let me just take off my stupid hat -- I figured it out. USER ERROR!!
Polycom apparently has you hit Transfer, the extension to send it to
(700) and send.
If you configure the polycom with the correct digitmap, then you will
never
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