Hi,
you can do print the dialstatus to the console e.g.:
exten = s,n,NoOp(${DIALSTATUS})
More info:
http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
Bye,
Patrick
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Hi,
at first: why do you use capitals for your name? Don't do that if you
don't have a very good reason.
You can convert wav to mp3 on the recording server and then send it to
the central system.
Bye,
Patrick
On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
Hello,
we are using vyatta, a linux based router. the software is more
focused on routing capabilities, than on firewall rules, but it works
fine an there is a very good support. for ha you can use it in a
cluster.
bye,
patrick
--
Niemann + Frey GmbH
Bischofstraße 80
47809 Krefeld
Tel. +49
Hi,
maybe you wan't to use '0' in front of you telephone number. eg.
intern: 261 - 261
exten: 002151-5462 - 021515462
Bye
On Tue, Aug 18, 2009 at 9:08 AM, Olivieroza-4...@myamail.com wrote:
Hi,
I need to replace digital handsets in offices where there cabling is
appareantly not
You can also use different identities.
On Tue, Aug 18, 2009 at 9:08 AM, Olivieroza-4...@myamail.com wrote:
Hi,
I need to replace digital handsets in offices where there cabling is
appareantly not Ethernet-compliant.
Today's usage is to press a key to toggle between private ou public line
Hi,
well there are differnt ways to do it. It depends on what you want.
The start-stop scripts in /etc/init.d/ are looking for a pid file, so
they can figure out if the server is running. You can change the
script to get a message if the server is going up or down by the
script.
If you want that
hi,
you can use call-limit=1 in sip.conf or DEVSTATE()
http://www.voip-info.org/wiki/view/Asterisk+func+device_State
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
http://www.voip-info.org/wiki/view/Asterisk+sip+incominglimit
bye
On Tue, Aug 18, 2009 at 9:03 AM, James
hi,
stunaddr = stun.exiga.net looks wrong ^^
in generally it looks like a nat problem.
bye,
patrick
On Mon, Aug 17, 2009 at 8:12 PM, Daniel Bareirodaniel-lis...@gmx.net wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm trying to connect to ekiga.net through a client
can also use a WLAN adapter to use usual sip phones without to many cables.
Bye
On Tue, Aug 18, 2009 at 1:20 PM, Olivieroza-4...@myamail.com wrote:
2009/8/18 Patrick Plattes patr...@erdbeere.net
You can also use different identities.
Yes, it's true but the trouble is to find a convenient way
hi,
try something like this:
bchan=4-5
hardhdlc=6
echocanceller=mg2,4-5
and then it should look like:
Asterisk-1:~# dahdi_cfg -v
DAHDI Tools Version - SVN-trunk-r6902
DAHDI Version: SVN-trunk-r6946
Echo Canceller(s): MG2
Configuration
==
SPAN 1: CCS/ AMI Build-out: 0 db
Hello,
I think there is a problem with chars in the extension name. I have a
similar issue if i try to use my my que management macro with a
extension with characters.
On Mon, Aug 10, 2009 at 3:16 PM, Tarek Sawahtareksa...@hotmail.com wrote:
i faced the same problem with callcentric.. when i
Hello,
i have a problem with the context parameter in the sip.conf. i'm using
a german sip provider (sipgate.de) and everything worked fine in
asterisk 1.4, but on 1.6.1 i got the following error message:
NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to
extension
Thanks for the fast reply, but it does not help :-(.
Bye, Patrick
On Mon, Aug 10, 2009 at 1:01 PM, Alex Balashovabalas...@evaristesys.com wrote:
Try prefix your extension in extensions.conf with _, e.g.
exten = _123,1,...
--
Sent from mobile device
What does dialplan show testing output?
[ Context 'testing' created by 'pbx_config' ]
'261' = 1. Noop(261) [SIP]
'262' = 1. Noop(262) [SIP]
'263' = 1. Noop(263)
have this in my Sipgate setup and it works. Worth a try.
Cheers
Andy
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick
Plattes
Sent: 10 August 2009 11:56
To: Asterisk Users Mailing List - Non
Hi Jonas,
that works fine, but I think its just a work arround and not a real
fix :-). For the moment it is okay and I'll try to fix the error next
days.
Thanks,
Patrick Plattes
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Does it this link help?
http://www.voip-info.org/wiki/view/Asterisk+cmd+MusicOnHold
On Fri, Aug 7, 2009 at 10:07 PM, Venkateshwarlu
Kakkirenivenka...@iconsultech.com wrote:
I want to a place a call (SIP) on hold in asterisk? Is there any way to do
it? If yes, please give me an example. We are
Hi,
Vyatta Asterisk works fine here. We are using traffic shaping DynDNS and NAT.
Bye,
Patrick
On Tue, Aug 4, 2009 at 2:27 PM, Barry L. Klineblkl...@attglobal.net wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tarek Sawah wrote:
First of all it acts like a firewall and a router..
Hello,
well let me explain one part of your question, the host parameter. if
you want to restrict the access to one ip you can say it here.
host=192.168.2.13 means, that you can only use this account from
192.168.0.13, eg. for security reasons. i recommend so set it to
dynamic at the moment and
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