Hello
Does anyone have any experience with configuring BT (British Telecom)
ISDN2e lines to work with Patton SmartNodes - and then Asterisk?
I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e
lines - and in turn connected to our internal LAN. I'm having huge
issues configurin
Hello
Does anyone have any experience with configuring BT (British Telecom)
ISDN2e lines to work with Patton SmartNodes?
I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e
lines - and in turn connected to our internal LAN. I'm having huge
issues configuring the SmartNode to s
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk dies when external access is lost
Phil Knighton wrote:
> Hello
>
> Looking for some help with a rather odd problem. We have Asterisk
> 1.4.10 running on a Linux box, within our Windows domain. Our Domain
>
lost
Phil Knighton schrieb:
> When we lose either our Domain Controller (for a reboot/maintenance)
> or external ADSL access, Asterisk drops all SIP registrations - even
> internal SIP calls within the building no longer function.
Not sure if it cures your problem but I would sug
simple in the
config, but I've been tinkering with this issue since we were using
Asterisk 1.2 and I've still not resolved it.
Any help or comments would be appreciated...
Thanks in advance
Phil
Phil Knighton
___
-- Bandwidth and Co
Hello all
What I'm looking for is some plain speaking advice on ISDN.
Currently using 4 analog lines connecting via a four port TDM400P FXO card. We
need to physically move our installations, and on requesting the analog lines
be moved - our telco (BT) is suggesting we replace our analog lin
Hello
I'm looking for a way to do the following using my Asterisk system and
Snom SIP phones...
Scenario:
Caller on Internal Phone 1 calls internal phone2. Phone 2 is busy (or
more accurately goes straight to voicemail).
Caller on internal phone 1 can press a button / dial a code (explained
2/Gradwell/44${EXTEN:1},,)
exten => _[1-9]X,1,Dial(IAX2/Gradwell/441353${EXTEN},,)
From: amit salunkhe [mailto:[EMAIL PROTECTED]
Sent: 14 February 2008 07:44
To: Phil Knighton
Subject: UK -999 dialing issue
HI Phil
Can u send me ur out call cont
Can I just say I'm grateful for all the replies - this list is
invaluable.
Thanks for the suggestion Razza, I've been back again to the logs and no
call was placed that contained the string "999" or "112" at the right
time!
Glad it made you smile, said it was a fun one for the list. Looking
li
D] On Behalf Of Tilghman
Lesher
Sent: 13 February 2008 15:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] UK issue - Asterisk dialling 999... sort
of
On Wednesday 13 February 2008 08:12:25 Phil Knighton wrote:
> Thing is, I have checked both our m
calls placed at about the time of the "999" calls?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil
Knighton
Sent: 13 February 2008 14:12
Hello
This is a fun one for the list...
d out.
There are no phones connected in anyway other than via Asterisk, fax
number is dealt with by a virtual machine, alarm system is on a
different number...
Any ideas before the rossers come and take me away?
Phil
Phil Knighton
Support Engineer
MJog Support Team
Soft O
Hi Adrian
I'm using 1.4.10, and all of my voicemail sound files are in SLN, so you
should be able to use them without a problem. Just check the notes for
the slightly different location for sound files under 1.4.
Thanks
Phil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PRO
28
To: Phil Knighton
Cc: asterisk-users@lists.digium.com
Subject: Re: Re: Lamps on Snom phones
Hello Phil,
please check the following details in your asterisk configuration and on
your phones. These are the settings that work for me:
sip.conf
[general]
limitonpeers=yes
allowsubscrib
to why.
-Original Message-
From: Stefan Guenther [mailto:[EMAIL PROTECTED]
Sent: 03 January 2008 12:21
To: Phil Knighton
Cc: asterisk-users@lists.digium.com
Subject: Re: Lamps on Snom phones
Phil Knighton schrieb:
> Thanks for your reply Stefan :-)
>
> I'm using Aste
hat next?
Phil
-Original Message-
From: Stefan Guenther [mailto:[EMAIL PROTECTED]
Sent: 02 January 2008 13:55
To: asterisk-users@lists.digium.com
Cc: Phil Knighton
Subject: Re: Lamps on Snom phones
Hi,
Phil Knighton wrote:
>Been through lots of stuff in the forums, and as far as I can t
I'm missing?
Any ideas would be appreciated!
Cheers
Phil Knighton
Support Engineer
MJog Support Team
Soft Option Technologies Ltd
The Old School, 23 High Street, Wilburton, Cambridgeshire, CB6 3RB
Tel: 01353 741641 | Fax: 01353 741341 | Email: [EMAIL PROTECTED]
Rilawich,
We use a TDM400P here in the UK, and if you set
"hanguponpolarityswitch=yes" in zapata.conf, we get the same result. I
think it is country specific, but try switching this to "no" and see
what happens. This cured our problems.
I have a note in zapata.conf (not sure if its from the rel
Hello
As a person who is somewhat a "newbie" to Asterisk, I have been given
the task of preparing our 1.2 installation for upgrade. The thing that
has slowed me down is some of the gaps in information on the upgrade
process. What's on the Wiki might make complete sense to both
experienced Linux
,1,Goto(so-switchboard,s,1)
exten => 5,1,Goto(so-silentdial),s,1)
exten => s,10,Background(softopt/som-mainmenuretry)
exten => s,11,Wait(1)
exten => s,12,Background(softopt/som-mainmenuopts)
exten => s,13,Goto(s,7)
exten => ,1,Playback(softopt/som-debug)
exten => 9999,2,Hangup()
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