?(LINEBUSY):(OTHER)
1,n(LINEBUSY), Wait(30)
1,n,goto(something,1,1)
1,n(OTHER), do something else
Sure it is pretty rough, but the basics are there. Also you might
want to read this: http://www.voip-info.org/wiki-Asterisk+variable
+DIALSTATUS
Kevin
Noah Silverman wrote:
Hi,
Does anybody have an
Hi,
Does anybody have an easy solution for this.
I want something that will keep trying a busy number every 30 seconds
until it gets through.
I've tried retrydial, but can't get it to work.
Any suggestions?
Thanks,
-N
___
--Bandwidth and Colocat
I have no idea. Whatever was the default when I set up the system
months ago...
-N
On Dec 8, 2005, at 2:36 PM, C F wrote:
What codec are you using?
On 12/8/05, Darrick Hartman <[EMAIL PROTECTED]> wrote:
Noah Silverman wrote:
Moj,
It is set as the default. *1
When I dial &
I have a related issue.
I have everything set up correctly so that I CAN use live recording
(Press *1 to start and stop recording.)
When I press *1, the console indicates "user pressed *1 to start
recording." I also hear the "beep" and an audio file is created.
The problem is that the aud
Thanks,
-N
On Dec 7, 2005, at 4:24 PM, Mojo with Horan & Company, LLC wrote:
Does your features.conf specify a custom setting for automon? If
it does, is that what you were dialing?
ie.
[featuremap]
automon => *#
Moj
Noah Silverman wrote:
Tried that,
Doesn't seem to do any
OK,
The plot thickens.
I've managed to get everything configured so that the system WILL
create a file. The problem is that the file just contains silence.
If I have a 10 second call that I record, I just get a wav file with
10 seconds of silence.
Anybody have an clues??
Thanks,
-N
Hi,
I tried setting verbose to 50 and never got any feedback on the CLI
about a pressed key...
-N
On Dec 7, 2005, at 3:53 PM, Time Bandit wrote:
That helps, but I'm still missing one piece.
I want to be able to press a button during the call to start and stop
recording.
I tried using:
Tried that,
Doesn't seem to do anything...
-N
On Dec 7, 2005, at 3:38 PM, Time Bandit wrote:
I'm trying to figure out how to setup "live" recording of a phone
call.
I've read all the docs at the wiki, but can't seem to figure out how
to implement it.
I'm running asterisk 1.2
I have the Po
Thanks,
That helps, but I'm still missing one piece.
I want to be able to press a button during the call to start and stop
recording.
I tried using:
exten => s,1,Dial(101,20,Ww)
But it doesn't seem to do anything.
-N
On Dec 7, 2005, at 3:29 PM, Philip Edelbrock wrote:
N
Hello,
I'm trying to figure out how to setup "live" recording of a phone call.
I've read all the docs at the wiki, but can't seem to figure out how
to implement it.
I'm running asterisk 1.2
I have the Polycom IP500 SIP phones.
In a perfect world, I would dial something to start recording, an
Hello,
I'm trying to figure out how to setup "live" recording of a phone call.
I've read all the docs at the wiki, but can't seem to figure out how
to implement it.
I'm running asterisk 1.2
I have the Polycom IP500 SIP phones.
In a perfect world, I would dial something to start recording, an
Great suggestion. I'll try it ASAP.
Where do I get fxotune?
Thanks!
-N
Matt Fredrickson wrote:
> On Tue, Apr 12, 2005 at 10:16:16AM -0700, Noah Silverman wrote:
>
>>I have a strange echo problem.
>>
>>When speaking on the phone with someone, I hear MY OWN vo
; box and the SIP phones. That's where the delay is coming from. You're
> not going to have significant jitter or delay problems on your local
> network, so adjust your jitter buffer down to 1. It will make your
> calls better from a latency point of view and it might help with
hat I can't help you with (I've
>> got lots of telecom experience, but little Asterisk experience) is
>> changing the settings in Asterisk to cancel it. The good news, though,
>> is that this is a straight-forward echo cancellation problem, and once
>> you find someone
s are Polycom IP500
7) I have the codec set to ulaw
Thanks!!!
-N
Jeff Heath wrote:
>On Tue, 2005-04-12 at 15:28, Noah Silverman wrote:
>
>
>>Hi,
>>
>>I tried, and still get an echo.
>>I don't think the problem is with the zap interface. It must be on the
>
Hi,
I tried, and still get an echo.
I don't think the problem is with the zap interface. It must be on the
asterisk or phone side.
-N
Rich Adamson wrote:
>>I have a strange echo problem.
>>
>>When speaking on the phone with someone, I hear MY OWN voice with a
>>sever echo. The other party so
I have a strange echo problem.
When speaking on the phone with someone, I hear MY OWN voice with a
sever echo. The other party sounds perfect, and they can hear me
perfectly. It is as if only the sidetone has an echo.
I'm running * on a dedicated box, small LAN, and am using 4 FXO cards to
con
Hi,
Somewhere in the Wiki I read that the best way to adjust the rxgain and
txgain is to dial a "type 102 milliwatt test line".
This line is usually found in xxx-958- or xxx-959- ranges.
I'm in area code 323 in Los Angeles.
Does anybody know the test number here??
Thanks,
-N
_
Thanks!
Robert Webb wrote:
>
> On Tue, 29 Mar 2005 12:30:31 -0800
> Noah Silverman <[EMAIL PROTECTED]> wrote:
>
>> hi,
>>
>> We are using PTSN lines connected through the Digium FXO modules for our
>> incomming lines
>>
>> When a c
hi,
We are using PTSN lines connected through the Digium FXO modules for our
incomming lines
When a caller calls in, the prompts play back at a really high volume.
They are a bit distored and fuzzy since they are so loud.
Can anybody give me some suggestions??
Thanks,
-N
_
rote:
On Mar 28, 2005, at 3:22 PM, Noah Silverman wrote:
Hi,
When someone calls into our * system over a PTSN line, we answer with
a recorded prompt. (Thank you for calling, etc..)
The first second of this prompt ALWAYS skips. After that, everything
sounds great and works perfectly. T
I tried inserting one second of silence before the first prompt. That
seems to work. You don't hear any "shop"
Kevin P. Fleming wrote:
Robert Goodyear wrote:
Anyone know if WAIT is not advisable to workaround the problem Noah's
asking about?
I always Wait(1) before answering an incoming PSTN
Thanks Rob. Let me know if you come up with anything.
Another option would be to ANSWER and then play one second of silence.
If there is "chop" during that second, nobody will notice.
-N
Robert Goodyear wrote:
On Mar 28, 2005, at 3:22 PM, Noah Silverman wrote:
Hi,
When someone call
Hi,
When someone calls into our * system over a PTSN line, we answer with a
recorded prompt. (Thank you for calling, etc..)
The first second of this prompt ALWAYS skips. After that, everything
sounds great and works perfectly. There is nothing wrong with the prompt.
Does anyobdy have any cl
The upgrade to the latest CVS-stable did the trick.
Now I have music on hold!!
-N
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Thanks Eric,
I downloaded the latest version of Asterisk about 4 days ago.
( I just got on the mailing about 3 days ago, so I couldn't have been
"following" it for long.)
-N
Eric Wieling aka ManxPower wrote:
Noah Silverman wrote:
How??
There is a nice big "hold" button o
How??
There is a nice big "hold" button on the phone. How do I re-configure
the IP500 so that * handles the hold???
-N
Steven Critchfield wrote:
On Sun, 2005-03-27 at 09:58 -0800, Noah Silverman wrote:
Hi,
I am having some trouble with music on hold.
Here is the situation.
Aster
Hi,
I am having some trouble with music on hold.
Here is the situation.
Asterisk Server. Polycom IP500 phone. Everything is configured and
works perfectly for incoming and outgoing calls.
1) If I use the hold button on the IP500 phone to place a caller on
hold, they just get silence.
2) I mad
e actual ID underneath it. Unfortunately, I don't think that this is
possible.
-N
C F wrote:
On Fri, 25 Mar 2005 16:06:53 -0800, Noah Silverman <[EMAIL PROTECTED]> wrote:
I have working with a polycom IP500 phone.
I like the idea of having each line button on the phone as a separate
sip dev
I have working with a polycom IP500 phone.
I like the idea of having each line button on the phone as a separate
sip device. If I understand it right, each phone could have three
extensions (one for each line.) This would be great since I could then
use the dialplan to forward calls to the des
eve you are the only one who suggested he use
[EMAIL PROTECTED] [EMAIL PROTECTED] is a great (actually stupendous) product
for those that want to have a PBX with a GUI up and running within a
few minutes. It does not, however, force users to really learn what's
going on underneath thi
ndous) product
for those that want to have a PBX with a GUI up and running within a few
minutes. It does not, however, force users to really learn what's going
on underneath things. Noah Silverman expressed an interest in learning
asterisk and its workings, so [EMAIL PROTECTED] is probably n
you to use
[EMAIL PROTECTED] why aren't you doing this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Silverman
Sent: Friday, March 25, 2005 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
Dean,
I'm not using [EMAIL PROTECTED]
What is "amp" and can I download it separately?
-N
dean collins wrote:
Noah you can, why not use amp (via [EMAIL PROTECTED]) to configure incoming
groups.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
We have two small business that run out of our office. One business has
3 phone lines, and the other has only one.
In a perfect world, Asterisk would indicate WHICH line (or group) the
outsider caller called, so that we would know which way to answer the
phone. The incoming calls would go thr
Hi,
I've managed to get my asterisk server up and running with a single POTS
line and a polycom IP500.
It will happily answer the phone line, tranfer calls, voicemail, etc.
The problem comes when I pick up the polycom phone and want to place an
outside call.
If I dial 913237773456 it just give
Randy,
I tried that already. Doesn't seem to help.
Thanks
-N
Randy Smith wrote:
My problem is that I can't seem to get the phone and Asterisk to
communicate with each other.
Any ideas??
-N
I have several of these phones. I had to upgrade my firmware. I'm running
1.4.1.0040 currently wit
y the phone won't register?
-N
Jason Brown wrote:
K. Now ere are the configs, minus the sip.ld file which is too big to
send to you. I recommend you have the latest 1.41 firmware.
Jason
-Original Message-----
From: Noah Silverman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 23, 2005
that I can't seem to get the phone and Asterisk to
communicate with each other.
Any ideas??
-N
Don Murray wrote:
Noah Silverman wrote:
Dean,
I appreciate the suggestion. Is it really necessary.
I've got slackware already installed on the box. (I consider myself a
bit of a Linux
ly.
Cheers
dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Silverman
Sent: Wednesday, March 23, 2005 6:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] polycom 500 help!!
Hi,
I'm just setting up my first Asterisk box. So far e
Hi,
I'm just setting up my first Asterisk box. So far everything is working
fine. I have the digium card in and connected to a regular telco line.
The Asterisk box answers the line and goes through the demo voicemail
functions. Sounds great!
I bought a Polylcom ip500 phone. I can't seem to
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