:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Panasonic Hybrid Integration Advice Needed
I disagree on this, you will have to create a dialplan in the
panasonic to tell it when to go over the ISDN circuit.
On 1/24/07, Scott Pinhorne [EMAIL PROTECTED
If you use a Vegastream gateway on the actual incoming ISDN circuits then
you won't even need to touch the Panasonic to integrate both systems.
Regards
Scott Pinhorne
VoxIT Limited
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: 24 January
Hi All
Is there an app or clever piece of dial plan that allows you to pull a call
back from voicemail when you missed it?
I am often on the phone and see I have another call, I hang up and the call
has already gone to voicemail, is there anyway to pull this back to my
phone? On our current
Hi All
Can anyone suggest a comprehensive GUI manager for Asterisk. It doesn't
matter if it is open source or commercial.
We currently have 100's of users currently managed via the real time
database. Groups of users belong to their own contexts.
We would like a system that is able to
Hi All
I am not sure what I wish to do it possible but I would like
to see if you guys know any better.
I have a site who has the extensions: 1231, 1232. 1233, 1234
Each of these users can dial each other on the extension
number an also has an external CLI mapped to them.
On all
Hi All
I am not sure what I wish to do it possible but I would like
to see if you guys know any better.
I have a site who has the extensions: 1231, 1232. 1233, 1234
Each of these users can dial each other on the extension
number an also has an external CLI mapped to them.
On all
Hi
Does anyone know how I can check if a callerID is more than
2 digits.
I am setting up my phones so that if the callerID is 3
digits the phones ring one way if it is more than 3 digits it rings another
i.e. internal calls and external calls.
exten = ,1,GotoIf($[${CALLERIDNUM}
Hi All
If this is the incorrect place or people can suggest a
better forum to post or an Asterisk consulting service I would be most
grateful.
I am using an SBC with our * server. All end points register
via the SBC to the * server.
The sip_buddies table has an entry for user
Hi All
I would greatly appreciate some advice or some direction as
to where to go next.
I have a provider passing me incoming calls via my Session
Border Controller.
I am able to pass them calls fine but coming in fails with a
407 Authentication Fail error.
In my sip.conf I have
Hi All
Would someone be kind enough to provide/point me to a resource when I
can see an example dialplan for making outgoing calls.
All our calls with go out via an ISDN30 gateway so ideally the diaplan
needs to be able to deal with the following errors:
no free channels
user busy
user
Hi All
I have 2 sip users setup in the database for realtime and they also have
their extension setup in the database.
When I register user 1 fine and can make and recieve calls.
As soon as i register user2 user1 is then unable to make any calls??
If i put the config fr both users in the
Hi All
Anyone have any knowledge of using the above?
I have installed it and wish to send out a fax via a SIP channel on an
ISDN3O. I have used the test script which says it has gone though ok but
i never see any activity in asterisk.
Checking the /var/spool/asterfax/tmp directory i can see
Hi All
I use Grandsteam GXP2000 phones.
Is there anyway within the dialplan/indications etc to have a custom
ringtone based on who is calling the phone.
i.e if i have a call from an internal user i get one ringtone if its an
external call i get a different ringtone??
Many Thanks in Advance
Hi
I wish to setup asterisk for training purposes so that I am able to
listen in to an extension while a call is going on?
Has anyone done this?
Thanks
SP
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Hi All
When a user wishes to 'break out' they prefix there call with a 9.
I know how to remove the 9 and then dial the remaining number and this
is all working fine.
I now need to remove the 9 but then prefix another number onto the phone
number before dialing now but am unsure how to do
Hi All
I have 2 phones registered to an asterisk server. The phones are sat
behind a NAT.
If I have the asterisk sat inline on the call after setting it up (with
transfer option specified as an example) the call works fine.
If I take out all options so the asterisk should bridge the call
Hi
I use a PIX 515 and had a similar problem when I started.
I turned on the fixup for SIP (as well as having nat in sip entry) and
it seems to do the trick for me.
Good Luck
SP
Bill Gibbs wrote:
Also the phone can dial out from behind the PIX…but obviously not
receive calls.
Bill
:-)
Many thanks
Scott Pinhorne
VoxIT.co.uk
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
in the queue?
Many Thanks In Advance
Scott Pinhorne
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Niklas
Thanks for this information I will be sure to follow it.
Many Thanks
Scott Pinhorne
Niklas Larsson wrote:
On Tue, 29 Nov 2005 06:14:54 +, scott wrote:
Is anyone using a vegastream product with asterisk? I have various
numbers coming into the vegastream vega400 and was after
20 matches
Mail list logo
