Displaying 20 results from an estimated 300 matches similar to: "Asterisk & X100P in Finland?"
2006 Feb 24
0
What's with Indications/SetLanguage/Zaptel/RingBack ?
Good morning everybody,
Can someone explain to me the interconnection between
these four things: indications.conf, SetLanguage(), zaptel.conf
and ring-back ? If there is any !! :- )
I am having this case where some users cannot hear ring back
from a DeadAGI script and it seems to be interconnected to these items.
These users are from the iaxfriends table, they _can_ hear ring-back from
a
2003 Nov 25
10
PCI 3.3 V
Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find
any motherboard with PCI 3.3 . Any sugestions!?
Cristian VASILIU
AccessNET International S.A.
Software Programmer
mail to :<cvasiliu@accessnet.ro>
www:<http://cvasiliu.home.ro>
2004 May 31
1
Where is my normal dialtone? With DLINK DG-104S (MGCP)
I once (for a brief period) had dialtone, but I do know why :)
Otherwise I get a boooop-booop-booop sequence.
I cannot tell if this is the D-Link doing this, or asterisk...
Who should be giving solid US dialtone?
My indication.conf says:
[general]
country=us
...
[us]
description = United States / North America
ringcadance = 2000,4000
2005 Dec 26
5
Asterisk Christmas Help request
Many thanks in advance for anyone that can offer help on the following
questions:
Asterisk Box
Using Asterisk@Home build and updated Asterisk to v2.1
P4, 400 Mhz, 384Mb RAM, 40Gb HD
4 OEM X100P Cards
Phones
Grandstream GXP-2000
2 * Grandstream BT-100
HandyTone 486
Sipura SPA-3000
Questions
1)
When someone calls in to one of the FXO lines, there is a 3-4 second delay
before the configured
2004 Apr 06
1
indications.conf settings for spain
Aqu? tienes,
[es]
description = Spain
ringcadence = 1500,3000
dial = 425
busy = 425/200,0/200
ring = 425/1500,0/3000
congestion = 425/200,0/200,425/200,0/200,425/200,0/600
callwaiting = 425/175,0/175,425/175,0/3500
dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425
record = 1400/500,0/15000
info = 950/330,0/1000
dialout = 500
Sergio Serrano Revuelto
Avanzada 7
Original Message:
2010 Jul 29
2
Disconnect supervision tone detection
Hi,
I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect
hangup tone or disconnect supervision tone from my CO. I attached the
recorded wav file which contains my telco's disconnect supervision.
I am using ,
asterisk-1.4.33.1
dahdi-linux-complete-2.3.0.1+
2.3.0
OS => Debian-lenny 5
users.conf
-------------
[trunk_1]
trunkname = pstn ; GUI
2008 Jun 06
1
Asterisk not picking up incoming calls from TDM400P
Hi,
I am having some issues with a new server install in Singapore.
Outbound calls work fine.
Inbound calls are not picked up by Asterisk.
Zaptel 1.2.25 and Asterisk 1.2.28 both built from source.
libpri installed
wctdm and zaptel load without error
Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface
Registered on major 196
Jun 6 23:34:03 fs01 kernel: [211138.372937]
2005 Aug 26
1
bridging sip to capi, no playtones back to caller
I've the following setup :
sip phone -> ser (auth and routing) -> asterisk with capi isdn
when I call a pstn number everything works fine, but I can't hear
anything till the called answer.
this is the output from a test call :
-- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack
-- Executing Dial("SIP/2.7.184.61-08152880",
2003 Jun 04
1
new application Dialtone()
Hello,
I created a new application for myself called Dialtone() by modifing
res/res_indications.c file. It can be used as such:
exten => s,4,Dialtone(30|${CALLERIDNUM})
exten => s,5,Playback(time-exceeded)
exten => s,6,Goto(s|1)
It will stutter if you have new voicemail and you have passed the mailbox
number as I did above. It will stop dialtone the moment you press a key
2005 Feb 02
0
ZAPHFC Drop calls
Hi everybody,
I had an ISDN card with winbond chipset and isdn4linux, but there was a lot of
echo when call from SIP to ISDN, so I buy an HFC chipset card (Conceptronic
c128i). I downloaded and compiled bristuff-0.2.0-RC5 and everything is going
fine. The sound quality is excellent, there is no echo, but i have a strange
problem. Sometimes calls made from SIP phones to ISDN just finished
2005 Jan 02
3
Indications UK - cant get away from american sounding dial tone
Have a problem which can't find solution to on WIKI..
Trying to get * to use UK based indication tones. i.e. british ring, dial
tone, busy signal.
Have changed the indications.conf file to default to UK. However this seems
to have no affect. What am i missing. Am using 1.0.3 stable.
Many thanks
Andrew.
----------------------
indications.conf
[general]
country=uk
[uk]
description =
2005 Jan 15
2
No sound with X100P (clone)
Hi, can't get X100P (fully zapata compatible clone) to work (I'm in
Australia).
* recognises the card and the channel (1) but has definetely some problems
talking to the pots line.
I set up this simple dialplan for ZAP ("incoming" context, as setup in
zapata.conf, for channel 1)
[incoming]
exten => s,1,Answer
exten => s,2,Playback(somefile)
exten =>
2004 Jul 14
1
Digium X100P card to a brazilian analog line
Hello,
I have a problem with connecting a Digium X100P card to a Brazilian analog
line.
Can somebody help me out with this problem?
My /etc/zaptel.conf is
loadzone=br
defaultzone=br
fxsks=1
My /etc/asterisk/indications.conf
[general]
country=br
[br]
description = Brazil
ringcadance = 1000,4000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/4000
congestion =
2010 Jun 18
0
Asterisk 1.4.33 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.33.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.33 resolves several issues reported by the
community, and would have not been possible without your participation.
Thank you!
The following are a few of the issues resolved by community
2010 Jun 18
0
Asterisk 1.4.33 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.33.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.33 resolves several issues reported by the
community, and would have not been possible without your participation.
Thank you!
The following are a few of the issues resolved by community
2003 Jun 12
0
Playtones unexpected hangups
1) I'm working on a quick replacement for DISA, and I ran into the
following snag: When I specify "Playtones(dial)" I can only get
around 7 seconds of wait time before the dialtone stops, and the
context goes to the "h" extension. Is there a way around this fixed
timeout? The DigitTimeout setting doesn't seem to have any effect at
all on this hangup problem. I
2004 Jan 21
0
Modulated tones not working
G'day list,
I have an Asterisk box built from very recent CVS (less than 24 hours
old), and I'm finding that the support for modulated tones (see
http://bugs.digium.com/bug_view_page.php?bug_id=0000746) is not working
for me for phones attached to my TDM400P.
I've set "country=au" in the [general] section of indications.conf, and
verified that the required changes have
2004 Jul 21
0
X100P only dials a single digit
Hey All,
Having a bit of a problem with a Wildcard X100P card. When I try to make
an outbound call using the card, it picks the line up and then only
dials a single digit. I've confirmed it's only dialling a single digit
by listening on a phone plugged into a parallel socket.
Incoming calls work fine, it's only outbound calls I'm having a problem
with.
I've tried
2005 Jan 15
0
X100P no sound problem
Hi, can't get X100P (fully zapata compatible clone) to work (I'm in
Australia).
* recognises the card and the channel (1) but has definetely some problems
talking to the pots line.
I set up this simple dialplan for ZAP ("incoming" context, as setup in
zapata.conf, for channel 1)
[incoming]
exten => s,1,Answer
exten => s,2,Playback(somefile)
exten =>
2005 Jan 16
0
X100P with no sound!
Hi, can't get X100P (fully zapata compatible clone) to work (I'm in
Australia).
* recognises the card and the channel (1) but has definetely some problems
talking to the pots line.
I set up this simple dialplan for ZAP ("incoming" context, as setup in
zapata.conf, for channel 1)
[incoming]
exten => s,1,Answer
exten => s,2,Playback(somefile)
exten =>