Hi,
You
should connect HG1500 with Ethernet port. But be careful, because in the HG1500
configuration, you need to declare nodes. This node should be the asterisk H323
[EMAIL PROTECTED] The asterisk need to be declare as a gateway, and not as a
phone!
I
never do such config, but as I was
Hello,
I had installed
several asterisk, but I every time had a problem with
callerID.
On each phones I
don't reveive the first digit.
For
example:
Caller 0672083516
called an IP Phone 0123456789. The IP Phone see 672083516 as
callerID.
I think there is a
patch for it, but I don't
17:37
À : Asterisk Users Mailing List - Non-Commercial Discussion
Cc : [EMAIL PROTECTED]
Objet : Re: [Asterisk-Users] Don't receive the prefix
On Tue, 4 Jan 2005, Eric Wieling aka ManxPower wrote:
GIBERT Frédéric wrote:
Hello,
I had installed several asterisk, but I every time had
Hi,
I had a question
regarding asterisk installation with 2 E100P card.
I would like to do
such installation but I had some troubles:
PABX - E100P
- asterisk - E100P - PSTN
In fact I would like
to add some IP Phones to my PABX wich is full.
But with this
installation, I can't place
Hello,
I'm actually trying
to connect an asterisk PBX with 2 E100P card to an alcatel 440, but I'm facing
some problems.
In fact, i had one
E100P connected to the public PSTN and the other one connected to the
Alcatel.
I can receive call
from the PSTN without any problems but I can't
, when I receive a call from my operator, if I can load balance
it on my 2 others E1 connected to the PABX.
I this case,
ifone PABX fail, I still had another one.
Thanks.
Regards.
GIBERT Frédéric Mobile: +33 (0) 6 7208 3516 Fax : +33 (0) 1 4692 0569
[EMAIL PROTECTED]
http://www.viginetworks.fr
Ste
will reach all
the previous explain path.
Is it possible to do
path replacement by clearing the channel between the 2 asterisk, and keeping the
voice between the telco, asterisk and the PBX?
Thanks.
Regards
GIBERT Frédéric Mobile: +33 (0) 6 7208 3516 Fax : +33 (0) 1 4692 0569
[EMAIL PROTECTED
to the phone port when you
receive a VoIP call?
Thanks.
GIBERT Frédéric
Mobile: +33 6 72 08 35 16
Fax : +33 1 30 71 39 33
Mail : [EMAIL PROTECTED]
Bureau Paris :
Ste VIGINETWORKS (Chez CAP retraite)
137, rue vielle du temple
75003 Paris
France
attachment
Title: Asterisk RC1 and bristuff
Hello,
Is the bristuff from junghanns.net are implemented in the asterisk RC1 release?
If no, is there a new patch from Junghanns in order the quadBRI card works?
Thanks by advance.
GIBERT Frédéric
Mobile: +33 6 72 08 35 16
Fax : +33 1 30 71 39 33
fax.
The calls between the 2 sites are in gsm codec. So the fax doesnt work!
Is there any possibilities to do caller based routing in asterisk, in order that when a fax try to send a fax, the call is automatically routed through the PSTN and not through the VoIP.
Thanks.
GIBERT Frédéric
Thanks for this informations.
Do you know where I can find the icd-snmp package for a redhat 9 distri?
I can't find it.
Thanks.
Message: 6
Date: Fri, 09 Jul 2004 15:45:57 +0200
From: Andrea Fino [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SNMP Monitoring
Reply-To:
.
GIBERT Frédéric
Ste VigiNetworks
Mobile: +33 6 72 08 35 16
Hello,
Is it possible to have 2 E100P cards on one asterisk?
I'm able to do that now, but I'm not sure about my config.
Here is my zaptel.conf
[EMAIL PROTECTED] asterisk]# more ../zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone=fr
defaultzone=fr
span=1,2,0,ccs,hdb3
Hi,
Is there a way to use swissvoice IP10 in MGCP mode behind NAT. I already
use this fonctionnality with a PBX other then asterisk, and it work very
well.
In fact, I need to use this settings in the IP10 telnet session
set xgcp rgw_name MAC_ADDR
set tcid 0 notify_entity [EMAIL PROTECTED]@IP
Hello,
Has someone know a good call generator for asterisk including SIP protocol
(freeware if possible)?
I need to stress a plateform and I don't find any.
Thanks by advance.
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]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Call generator
Reply-To: [EMAIL PROTECTED]
Andrew Kohlsmith wrote:
On Wednesday 23 June 2004 04:46, GIBERT Frédéric wrote:
Has someone know a good call generator for asterisk including SIP protocol
(freeware if possible)?
I need to stress
Hello,
I would like to know if someone gets a doc which resumes what changes need
a reload and what changes need a restart of asterisk.
Thanks.
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