I remember a thread similar to this a while ago but couldn't find. How do I
make Asterisk to interact with an IVR? (Nothing fancy, just plain
predictable voice menus like a conference bridge.) I get stuck at Dial(),
which seems to wait for hangup after the other end picks up.
Yuan Liu
Uh maybe not... :) stop now will cause Asterisk to drop *all* calls
and exit immediately. Kinda the equivalent of nuking a small city to
kill one person.
Paul Hales wrote:
Stop now?
PaulH
On Sat, 2007-02-03 at 21:47 -0800, Yuan LIU wrote:
Can I disconnect an arbitrary call using a
Yuan LIU wrote:
I remember a thread similar to this a while ago but couldn't find.
How do I make Asterisk to interact with an IVR? (Nothing fancy, just
plain predictable voice menus like a conference bridge.) I get stuck
at Dial(), which seems to wait for hangup after the other end picks up.
My TDM400/zaptel 1.2.10/Asterisk 1.2.13 suddenly stopped bridging outgoing
FXO calls. If I make a call (from an FXS channel) to a PSTN destination,
and the other side answers, Asterisk will show continued ringback on the FXS
channel, while the PSTN side hears silence. No error message
From: Leo Ann Boon [EMAIL PROTECTED]
Yuan LIU wrote:
I remember a thread similar to this a while ago but couldn't find. How do
I make Asterisk to interact with an IVR? (Nothing fancy, just plain
predictable voice menus like a conference bridge.) I get stuck at Dial(),
which seems to wait
From: Yuan LIU [EMAIL PROTECTED]
My TDM400/zaptel 1.2.10/Asterisk 1.2.13 suddenly stopped bridging outgoing
FXO calls. If I make a call (from an FXS channel) to a PSTN destination,
and the other side answers, Asterisk will show continued ringback on the
FXS channel, while the PSTN side hears
We have 2 line PBX office that connect to my VoIP Network that contain 2
Asterisk Server each server has FXO Digium. When I make call PBX-to-PBX
via VoIP Network and when done my FXO port still active
||
||
|PBX|-|Asterisk
Hi All,
How to install bristuff on asterisk 1.2.14? install scripts are trying to
download and compile those versions:
asterisk-1.0.10
zaptel-1.0.10
libpri-1.0.9
and I'm running:
asterisk-1.2.14
zaptel-1.2.12
libpri-1.2.4
I only need Pickup application from bristuff to be able to pickup
On Sun, Feb 04, 2007 at 11:24:36AM +0200, Dominik Zalewski wrote:
Hi All,
How to install bristuff on asterisk 1.2.14? i
You don't. You install bristuff 0.3.0-PRE-1x .
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406
I'm specing out a new box to act as a tandem switch. It will have a TE410P
with 4 x PRI and support IAX connections to four other boxes using
predominantly ilbc and/or gsm. It also has 3 IAX trunks to Teliax for call
routing also using gsm. No extensions actually terminate on the tandem,
All,
I'm haveing a bit of trouble getting my head around H.323 and call routing with
Gatekeepers, Zones and intra-zone calls - hopefully someone who is more
informed in things H.323 will be able to point me in the right direction...?
I already have a mature network of Asterisk boxes dotted
Last question for the day, I promise.
On voip-info.org and trixbox.org, I found some old threads on MWI via SMDI.
Has this been rolled into Tbox or has anyone successfully rolled it in after
the fact. As part of our longterm plan, I'd like to move the legacy PBX to
Tbox and pass MWI back to it
Well, it's like trying to check your hotmail.com email account from
netscape.net - it just isn't going to work.
What you can do, however, is talk to people on google talk from other jabber
systems, just like you can send a netscape.net user an email from hotmail.com
- Ian Hailey [EMAIL
Yuan LIU wrote:
From: Leo Ann Boon [EMAIL PROTECTED]
Yuan LIU wrote:
I remember a thread similar to this a while ago but couldn't find.
How do I make Asterisk to interact with an IVR? (Nothing fancy, just
plain predictable voice menus like a conference bridge.) I get stuck
at Dial(), which
Hello,
I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I
would like use analogue lines for outboud calls.
How is it possibile to detect ANSWER?
- answeronpolarityswitch does not seem to work in Italy
- call progress does not give safe results, sometimes calls get
billed,
On Sun, Feb 04, 2007 at 11:54:14AM +0200, Tzafrir Cohen wrote:
On Sun, Feb 04, 2007 at 11:24:36AM +0200, Dominik Zalewski wrote:
Hi All,
How to install bristuff on asterisk 1.2.14? i
You don't. You install bristuff 0.3.0-PRE-1x .
BTW: consider also asking on
Mochamad Susantok wrote:
We have 2 line PBX office that connect to my VoIP Network that contain 2
Asterisk Server each server has FXO Digium. When I make call PBX-to-PBX
via VoIP Network and when done my FXO port still active
||
||
Tzafrir Cohen already mentioned it in a reply to someone
else:
There's a bristuff-users mailinglist now.
If you are interested in bristuff or are using it consider
subscribing to it.
The webinterface is here:
http://lists.three-dimensional.net/mailman/listinfo/bristuff-users
The list address
Hi,
On Sun, 2007-02-04 at 16:17 +0100, Stefano Corsi wrote:
Hello,
I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I
would like use analogue lines for outboud calls.
How is it possibile to detect ANSWER?
you cannot. it's analogue, no signalling is done on it.
unless
I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I
would like use analogue lines for outboud calls.
How is it possibile to detect ANSWER?
you cannot. it's analogue, no signalling is done on it.
unless you write dsp routines to detect the right things
at the right moment
Indeed. The problem was the ).
thanks to all who helped me debug this...my eyes are not so young anymore...
On 2/3/07, jacobso1 [EMAIL PROTECTED] wrote:
hi,
i think the problem is here :
exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to)
|
replace with
That's a very good news !!
Congratulations to Tzafrir for it !
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Stefano Corsi wrote:
I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I
would like use analogue lines for outboud calls.
How is it possibile to detect ANSWER?
you cannot. it's analogue, no signalling is done on it.
unless you write dsp routines to detect the right things
I have the following dialplan (segment) that isn't working as I expected
it to:
exten = s,n,Dial(Zap/1SIP/202SIP/203,18)
exten = s,n,Dial(Zap/1SIP/201SIP/202SIP/203,42)
The plan was to have SIP/201 added to the group of ringing phones after
3 or so rings. What ends up happening, though, is
Hi,
I turned on dnsmgr in 1.2 and it worked for a few weeks. Suddenly, no
iax2 providers were working. All of them were unreachable. My own
fixed ip phones were. I disabled the dnsmgr and now the IAX providers
are working again. No big deal, but it's odd that this happened this
way. Anyone else
Yuan LIU wrote:
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
I remember a thread similar to this a while ago but couldn't find.
How do I make Asterisk to interact with an IVR? (Nothing fancy,
just plain predictable voice menus like a conference bridge.) I
get stuck at Dial(), which
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
Stefano Corsi wrote:
I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I
would like use analogue lines for outboud calls.
How is it possibile to detect ANSWER?
you cannot. it's analogue, no signalling is done on it.
unless
What PBX do you have connected to the Asterisk servers? My experience is that
most PBXs do not provide a disconnect signal on their analog station ports. I
have had the most success with disconnects on Avaya PBXs. Nortel analog
stations last I tested with any, did not provide a disconnect
From: Scott Walde [EMAIL PROTECTED]
I have the following dialplan (segment) that isn't working as I expected it
to:
exten = s,n,Dial(Zap/1SIP/202SIP/203,18)
exten = s,n,Dial(Zap/1SIP/201SIP/202SIP/203,42)
Interestingly, although the Asterisk Manual (by Mark Spencer and so on)
contains an
I define
[globals]
myvar = ${DB(store/myvar)}
---
But when I want to use ${myvar} in the dial plan, I found that the variable
is null when Asterisk is restarted. Only a reload would force globals to
read AstDB. Other variables in globals loads fine.
Any idea? (Asterisk 1.2.13)
Yuan Liu
Is there anything that allows a logical line to extend to the next physical
line? Printed files are so hard to read with blind line wraps - and my
printer doesn't even automatically wrap.
Yuan Liu
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how would a line be soo loogg?
On 2/4/07, Yuan LIU [EMAIL PROTECTED] wrote:
Is there anything that allows a logical line to extend to the next physical
line? Printed files are so hard to read with blind line wraps - and my
printer doesn't even automatically
On Fri, 2 Feb 2007 17:56:26 -0500
Wes Baehr [EMAIL PROTECTED] wrote:
The problem can be reproduced in the same way by putting a caller on
hold, unholding, and holding again. The MOH restarts from the
beginning of whichever file it was playing last. (I have random
enabled, so it randomly picks
Hi,
Out ITSP has told us to user SIP privacy headers to hide outbound caller
ID. Does anyone know how or if this can be done in Asterisk. I tried
exten = s,3,SIPAddHeader(privacy=on)
prior to executing Dial but no luck.
___
--Bandwidth and
Look here:
http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-
ID+header
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Bishop
Sent: Sunday, February 04, 2007 15:43
To: Asterisk Users Mailing List - Non-Commercial
Im glad to let you know that finally I invested some time to make work
Unicall in Asterisk 1.4, I must say not much testing could be done
since I have no hardware available ( cards, servers ), however a
friend was able to test it with a couple of calls with success, I need
you to test this and
thanks for that. Do you know what P-Asserted-Identity needs to be set to to
hide caller ID via privacy headers?
On 2/5/07, Darryl Dunkin [EMAIL PROTECTED] wrote:
Look here:
http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header
--
Yuan LIU wrote:
exten = s,n,Dial(Zap/1SIP/202SIP/203,18)
exten = s,n,Dial(Zap/1SIP/201SIP/202SIP/203,42)
Interestingly, although the Asterisk Manual (by Mark Spencer and so
on) contains an almost identical sample plan fragment, in reality, it
seems to need a Wait() in between to reset the
I had my setup working properly under 1.2 and after a disk crash I
decided that I wanted to try Asterisk 1.4. So far I can transfer
between phones and I can dial out. What I can't get working is to
get an SPA-3102 to 'send the calls' to Asterisk. I have the device
added to the sip.conf file and
Hi everyone:
I'm trying to compile Asterisk on FreeBSD 6.0-RELEASE and I'm getting the
following error:
cc -O2 -fno-strict-aliasing -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -Iinclude -I../include
-D_REENTRANT -D_GNU_SOURCE -DMAKE_VALGRIND_HAPPY
I am don't sure what PBX we have, but when i hangup i hear like tut tut
tut in the other side, Is that not disconnected signal ?
FYI when i take call from PBX to VoIP client or vice versa. it's ok. What
do you think, are there is have some relation with my problem ?
What PBX do you have
This would do it, but a better way would be to specify --with-zaptel=PATH
(PATH is the directory of zaptel sources) when running configure. If you
already did a build you probably want to run make dist-clean before running
configure again.
Best regards,
Alex
Alex Epshteyn
Third Lane
Hi All,
I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to
this page
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ;
when I dial ,there have this warning:
-- Executing AsyncGoto(SIP/111-086497c8,
SIP/113-08674628|dynamic-nway|111|1) in new
Hi All,
I use the Asterisk Manager Interface to redirect the channels.
There have two channels :
SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456)
SIP/612-5456 [EMAIL PROTECTED]:10 Up Dial(SIP/[EMAIL
PROTECTED]
Then I send a
Hello,
We will be having another Tampa Bay area Asterisk User Group meeting
on Monday, February 5th at 7PM
All Asterisk users from newbies to Gurus are encouraged to attend.
For more information visit our website:
http://asteriskpbx.meetup.com/1/calendar/5394922/
Thanks,
MATT---
As everybody must be watching the superbowl. I post this to let you
have some fun while thinking what this can be.
TDM400p (fxo) connected via loopstart to ports in an AvayaG3
call comes in from the avaya to the tdm card:
WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with
error
Steve, keep me in touch please ?
We are also looking for moving all our activities to java platform.
Let me know if You'll find/test something like asterisk2billing written in
java ?
Cheers,
Kate
On 2/1/07, Steve Prior [EMAIL PROTECTED] wrote:
When I was looking for a Java FastAGI interface
I have a weird problem
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext - Panasonic Ext No Problems
Panasonic Ext - SIP Ext No Problems
SIP Ext - VOIP Provider No Problems
Panasonic Ext - VOIP Provider Errors
-- Working SIP -
No, that is just a tone. Correct disconnect supervision is an
electrical thing. Either reversing the polarity or dropping battery.
Mochamad Susantok wrote:
I am don't sure what PBX we have, but when i hangup i hear like tut tut
tut in the other side, Is that not disconnected signal ?
FYI
From: Scott Walde [EMAIL PROTECTED]
Yuan LIU wrote:
exten = s,n,Dial(Zap/1SIP/202SIP/203,18)
exten = s,n,Dial(Zap/1SIP/201SIP/202SIP/203,42)
Interestingly, although the Asterisk Manual (by Mark Spencer and so on)
contains an almost identical sample plan fragment, in reality, it seems to
Kate Kretz wrote:
Steve, keep me in touch please ?
We are also looking for moving all our activities to java platform.
Let me know if You'll find/test something like asterisk2billing written
in java ?
I haven't received any feedback at all on the relative use of the java
options, but I'm
From: C F [EMAIL PROTECTED]
how would a line be soo loogg?
It doesn't take a very complicated expression to go over 80 characters.
Also consider multiple voice files in PlayBack() or Background(), System()
calls, etc.
Yuan Liu
On 2/4/07, Yuan LIU
Hi all,
I am preparing the new asterisk system for 60 concurrent calls with 2 E1.
I have to use server HP DL380 G5.
Anybody get TE212P card work on this server using asterisk?
Thanks,
M
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Very odd.
My guess is that it's one of 2 things.
Slightly different number being sent to the SIP provider. (unlikely)
Different callerid being sent to the SIP provider.
Have you tried blanking the callerid before making the outbound call?
(in case the provider doesn't like it)
PaulH
On Mon,
Another dumb question: Can a dial plan continue after local hangup when
using Dial()? For example,
[incoming]
exten = s,1,Dial(Zap/1)
exten = s,2,Congestion()
exten = s,3,Hangup()
---
Asterisk seems to insist that a dial plan is complete when Zap/1 hangs up
and do not go into priorities 2 and
Yuan LIU wrote:
Another dumb question: Can a dial plan continue after local hangup
when using Dial()? For example,
[incoming]
exten = s,1,Dial(Zap/1)
exten = s,2,Congestion()
exten = s,3,Hangup()
---
Asterisk seems to insist that a dial plan is complete when Zap/1 hangs
up and do not go into
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