Hello.
I am trying to
connect Asterisk to Cisco Gatekeeper running on 2600 router. On the
Gatekeeper I get the loopback address from Asterisk in "CallSignalAddr".
All Other devices that registers on this Gatekeeper report their ethernet ip
address in this section.
router1#show
gatekeeper
Hi Prasad,
Install a Asterisk in your DMZ and one Asterisk inside of your Lan.
Set them to use IAX between them passing through your firewall.
A) Your SIP Phones in your lan will connect to your LAN's *.
B) The SIP Phones in the internet will connect to your DMZ's *.
C) A connnects to B through
I got a strange error message on the CLI, saying:
Warning: File.c:550 ast_readaudio_callback: Failed to write frame
any ideas ?
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* Ian D. Willoughby (Wed, 3 Nov 2004 10:49:44 +)
Are you based in Hastings by any chance (Senlac and all that)?
Heh. No, Senlac is my street name in Romsey, Hampshire.
I
On Wed, 3 Nov 2004 13:10:57 +, [EMAIL PROTECTED] wrote:
On Tue, Nov 02, 2004 at 10:58:39PM +, StrUK wrote:
snip other information
I guess my question is: does anyone have polarity reversal hangup
detection working on a BT line with an fxo module in a TDM400P?
Testing with my fxo
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Walt Reed
Sent: Wednesday, November 03, 2004 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Reject a call if no callerID
On Wed, Nov 03,
But now that logic works. However how would you insert that into the
dialplan to get it to work or would AGI be better solution?
Brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Wednesday, November 03, 2004 3:25 AM
To:
steve szmidt wrote:
On Tuesday 02 November 2004 08:18 pm, Julio Arruda wrote:
Steve Underwood wrote:
'replace' fax over G.711 by fax over G.711 over IP :-)
The point being, Fax over VOIP (even using G.711), I don't believe would
be as reliable as Fax over an ISDN b-channel :-) Better
Hello all,
The asterisk-oh323 package has been updated. From now on, there are two
series of releases:
- 0.6.x releases, latest is 0.6.4. These will work with Asterisk v1-0
source code.
- 0.7.x and above, latest is 0.7.0. These are for CVS code of Asterisk.
Also, the latest versions now use
Michael
The problem lies not with Asterisk or the cards but with the combination
of voip and analogue telephony in general. I can guarantee that you
will have very similar echo when you connect your Panasonic pbx to the
analogue lines. In fact, your echo will most likely be worse. It is
just
On Wed, 3 Nov 2004 11:11:42 -0500, Christopher TenHarmsel
[EMAIL PROTECTED] wrote:
At the place I work we're using Asterisk to run our in-office phone
system. We have about 15 employees and a total of about 5 hard phones.
Right now when asterisk receives an incoming call, it rings all 5
Hello list,
I am trying to install a DigiumX100P into a Redhat Asterisk.Kernel seems to be OK, card OK.Zaptel Configuration seems to be OK.
# ztcfg -vvChannel map:Channel 01: FXS Kewlstart (Default) (Slaves: 01)1 channels configured.Asterisk works fine with IP SIP but not with X100PI get the error
I would like to know if there is a way to change default ulaw for a T1
card. I see in the zap show channel X that is working as ulaw. How do I
change it in zapata.conf or zaptel.conf to alaw. Iam interconnecting a
Meridian PBX but I need to configure it as alaw.
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