This is a UK site, but not sure if it's available elsewhere, or even if
it's what you want!
http://www.voiptalk.org/products/COM-ON-AIR+Desktop+Set+Clearance
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid
Bender
Sent: 22 January 2006 18:25
To:
Hi,
Im using Asterisk @ Home with 8 ISDN channels, about
30 extensions and a bunch of Cisco phones.
I was wondering if there is any way of mapping
an extension to a specific outbound trunk?
For example, there are 2 organisations in my building, 4
ISDN channels have one telephone
Hi,
I'm trying to get Asterisk working with the AVM B1 card. I've tried every
instruction set I can find, but to no avail.
I think I'm getting mixed up as to what Zaptel/CAPI configuration to use.
If someone is currently using one of these cards, would it be possible to
mail me a few bits of
analogue lines or 1 ISDN. 2
analogues would be cheaper I believe, what sort of ISDN card works well with
Asterisk?
Thanks for any help!
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Hi,
I'm looking for a reliable provider to use mainly for outgoing calls in the
UK, incoming isn't so much of a worry as I think I'm going to accept them
over ISDN.
Cheers!
Steve
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to be pressed to link Staff 3 with the external
call.
Has anyone done this or knows how to do it, or something similar?
Cheers,
Steve
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environment.
Thanks,
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about 30 seconds.
Until yesterday
it worked fine. I am using TE410P with 2 E1 connected
trunks
with h.323,
sip and skinny phones on voip side.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Steve Foy
Sent: Monday, February 09
, a call disappeared, and gradually got more more frequent.
Nothing appears in logs or console.
What phones are you using?
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setup. I would guess it is the
ACK that is not receiving a STATUS 200 OK so Asterisk cuts off the call.
No, they drop at random points in the calls. Sometimes after 30 seconds,
sometimes up to 5 minutes :(
Steve
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pissed :/
No more ideas?
I've tried everything else people have mentioned.
Cheers,
Steve
On Mon, Feb 02, 2004 at 01:03:01PM -0500, Bill Hamel wrote:
Hi,
Have you checked for IRQ conflicts ?
-b
Quoting Steve Foy [EMAIL PROTECTED]:
Hi,
On Mon, Feb 02, 2004 at 06:04:40PM +0100
have mentioned.
Cheers,
Steve
On Mon, Feb 02, 2004 at 01:03:01PM -0500, Bill Hamel wrote:
Hi,
Have you checked for IRQ conflicts ?
-b
Quoting Steve Foy [EMAIL PROTECTED]:
Hi,
On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote
,
Steve
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This strikes me as something that should be really very simple to do, but I
can't figure it out.
Is there a way of logging all SIP debuging info to a file somewhere?
It would help me greatly!
Cheers,
Steve
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Still no luck, calls are still dropping off about the same amount as before.
Any more ideas!?
On Fri, Jan 30, 2004 at 05:14:27PM +, Steve Foy wrote:
Thanks, I'll try that and see how it goes.
Cheers,
Steve
On Fri, Jan 30, 2004 at 11:46:05AM -0500, Bill Hamel wrote:
Try adding
heavily loaded or
anything like that...
Grr..
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running asterisk isn't under heavy load, so I can't see why this is
happening.
I am not using g.729 or 723, just plain old ulaw, which I have got enabled in
sip.conf
Cheers,
Steve
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callerid=Outside Line 2
channel=2
callerid=Outside Line 3
channel=3
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in the logs and nothing on the console, the call just seems
to 'go away'!
Can anyone shed any light on this?
Regards,
Steve
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to me.
Is there a way I can get the sip debug lines to get piped out to a file with
timestamps?
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secret=xxx
host=dynamic
dtmfmode=rfc2833
callerid=Shirley O'Neill 100
context=internal
[EMAIL PROTECTED]
qualify=yes
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callerid=Shirley O'Neill 100
context=internal
[EMAIL PROTECTED]
qualify=yes
canreinvite=no
-b
Quoting Steve Foy [EMAIL PROTECTED]:
Bill,
On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote:
Shot in the dark here ...
Do you have
GMT.
That's over 12 hours. My box has been up and accepting mail all night as it
does every night, so it's not a problem there.
Has anyone any suggestions or thoughts about how to improve it?
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' causes Zapateller to only play the tone if there
is no callerid information available. Options should be separated by |
characters
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else experienced this or have any suggestions as to how I could
fix it!?
Cheers :)
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correctly
when called normally.
Regards,
Steve
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before
playing the sound.
Did you do Answer before you call Playback() in your extensions.conf?
Yep, the extension looks like this:
exten = 502,1,Answer
exten = 502,2,Playback(nagios/hostdown)
exten = 502,3,Hangup
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Hi there,
I'm wondering if there is a way to assign a different Caller ID to each Zap
interface.
I have 3 Digium X100P cards, and I'm sure there must be some way of
configuring zapata.conf to allow each line to identify itself with a
different Caller ID string.
Many thanks,
Steve
--
Steve Foy
Apologies, I've got it to work.
I didn't realise by just specifying the channels individually and resetting
the Caller ID before each channel would work.
Regards,
Steve
On Mon, Jan 19, 2004 at 03:59:54PM +, Steve Foy wrote:
Hi there,
I'm wondering if there is a way to assign a different
sense!?
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