Yesterday I upgraded my Zaptel to 1.2.26 or I think that was it, the
latest 1.2 version at downloads.digium.com. I have a Digium 4 card
populated with 4 red FXO cards using channels 1,2 and 4. Channel 3 is
not used. It's been working fine for a few years. After upgrading to
1.2.26 calls
On Sun, Jul 13, 2008 at 12:05 AM, Steve Edwards
[EMAIL PROTECTED] wrote:
On Sat, 12 Jul 2008, Douglas Garstang wrote:
The person I am working is building a calling card. They want to allow
the user to recharge their account when their time runs out (without
hanging up the current call). I got
Hi,
I have problem using Asterisk.I have isdn-pri and openvox d110p card in my
computer.They are connected with RJ-45 (1,2,4,5 pins to the card and all
pins to the isdn done by telco workers). I got green led on isdn which is
sign that isdn is working and that is connected to openvox, right ? I
Skip2pbx ? They claim to convert ip to Skype but has a nasty price.
2008/7/13 Michael Graves [EMAIL PROTECTED]:
On Sat, 12 Jul 2008 10:54:07 -0400, Julio Arruda wrote:
Jason Aarons (US) wrote:
My understanding is Skype's secret is using the iLBC codec, which Cisco
has also licensed for
Hi,
I'm having strange warning from asterisk when I try to dial GSM Gateway:
-- Executing [EMAIL PROTECTED]:1] NoCDR(SIP/ibm-b2c52848, ) in
new stack
-- Executing [EMAIL PROTECTED]:2] Dial(SIP/ibm-b2c52848,
Zap/R3/501522xxx) in new stack
-- Requested transfer capability: 0x00 -
Hello,
We are proudly to present new version of our billing and routing system MOR
v0.6
More info: http://www.voip-info.org/wiki/view/MOR
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
___
-- Bandwidth and Colocation Provided by
I have a wildcard 100 xp on my pots line and all was working just fine
up until a few days ago when all of a sudden it stopped receiving caller
id on incoming calls. I know caller id is being presented on the line
as the analog set on the same line always gets it.
What is strange is that this
Hi, this is my first post to the list, but I have tried to search
elsewhere for a solution
SNIP
I'm using sipgate.co.uk for incoming calls, but when I make a test
call from the PSTN, the call just dies without connecting to my
Astlinux box. (I'm monitoring asterisk console via 'asterisk
Hey Guys,
I have configured my first asterisk box. it works ok so apart, but the
playback sound quality is terrible, its low and the output sounds distorted
and its seems to have been clipped.
Can anyone help.
On Sun, Jul 13, 2008 at 11:00 AM, Chris Rowson [EMAIL PROTECTED]
wrote:
Hi,
On Sun, Jul 13, 2008 at 6:45 PM, Leotis buchanan
[EMAIL PROTECTED] wrote:
Hey Guys,
I have configured my first asterisk box. it works ok so apart, but the
playback sound quality is terrible, its low and the output sounds distorted
and its seems to have been clipped.
Tried this?
Hi,
I think the important error message is jumping out of macro
'nway-conf-start' not ast_bridge_call.
It is because it is not allow to jump to another context when you use macro.
Best regards,
Charles
2007/4/23 Manu Mehta [EMAIL PROTECTED]:
Hi,
I am trying to achieve 3-way conferencing
Asterisk 1.4.xx does not show the Zap channel instance:
Started three way call on channel 25
-- Starting simple switch on 'Zap/25-1'
Asterisk 1.2.xx would show:
Started three way call on channel 25
-- Starting simple switch on 'Zap/25-2'
1.4.21.1 core show channels:
Zap/25-1
Hi all,
I have added a language pack nz under the sounds folder
I have changed zaptel.conf to be loadzone=nz defaultzone=nz
I have also changed the language in freepbx to New Zealand however when
I go to my voicemail on my phone the CLI gives me
SIP/604-094eaac0 Playing 'vm-password' (language
Dear All,
I have a problem compiling Zaptel on an up to date CentOS 5.2 box.
Zaptel 1.4.11, CentOS running on AMD dual core X64.
The configuration step finishes, but during the 'make' step it stops
here:
...
CC [M] /projects/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o
In file included
from
Try adding the following to your voicemail.conf context:
format=wav49|wav
Steve
Leotis buchanan wrote:
Hey Guys,
I have configured my first asterisk box. it works ok so apart, but the
playback sound quality is terrible, its low and the output sounds
distorted and its seems to have been
Hi Bob -
I have a problem compiling Zaptel on an up to date CentOS 5.2 box.
Zaptel 1.4.11, CentOS running on AMD dual core X64.
...
CC [M] /projects/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o
In file included
from /projects/asterisk/zaptel-1.4.11/kernel/xpp/xpd.h:26,
from
You should probably avoid giving incoming access to outgoing..
PaulH
Lee, John (Sydney) wrote:
With an ISDN10/20/30/etc, I would just put all the lines into an
'incoming' context - and make sure that incoming context doesn't have
any includes (unless you really need them...)
Can
You should probably avoid giving incoming access to outgoing..
Thanks Paul.
[incoming]
...
include = internal
include = outgoing
The thing is if I don't have this include = outgoing in [incoming], I
will not be able to dial out at all.
Any thoughts?
Matt Watson wrote:
I'd probably be a little pissed if I were Steve Underwood if somebody
pocketed over 10k $USD for taking credit for a product that my free library
did the bulk of the work for.
I can't speak for Steve at all, but any major contributor to an
open-source project faces this, so
You should probably look at having another context - maybe even
'sip-phones' for your sip phones.
Then include everything you need there.
PaulH
Lee, John (Sydney) wrote:
You should probably avoid giving incoming access to outgoing..
Thanks Paul.
[incoming]
...
include =
You really want to avoid people making incoming calls being able to make
outgoing calls.
Especially international ones.
PaulH
Lee, John (Sydney) wrote:
You should probably avoid giving incoming access to outgoing..
Thanks Paul.
[incoming]
...
include = internal
include =
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