If the version either 1.4.x or 1.6.x, run make menuconfig from
asterisk source directory, pick up sounds you need, and make install
then.
Or, you can do as mentioned above: install sounds you need explicitly.
On Mon, May 24, 2010 at 5:42 AM, ayodele abejide
ayodeleabej...@hotmail.com wrote:
hi,
Hello,
Can you explain how to test blind transfer in asterisk.
Here is my test case that hasn't succeeded:
I have configured blindxfer = # in features.conf. I have called an iax user
from my iax softphone. The called party responds to the call, and tries to
transfer the call by clicking the
Hello,
I have a problem, and I'm looking for you help.
When I dial certain number my calls are getting dropped.
I initiate the call, I hear IVR, then I am being transfered to
operator, and then suddenly I get ISDN DISCONNECT message.
I had this type of problem some time ago, and I thought it
Hi,
I'm expiriencing very annoying issue with Asterisk while using ODBC
on Linux box running Debian with following packages:
Asterisk-node0:/tmp# dpkg -l | grep -i odbc
ii libiodbc23.52.6-2
iODBC Driver Manager
ii libmyodbc
I have the Asterisk 1.6.2.x and it don't add reason header in the CANCEL!
Still no idea?
François
Le 21/05/2010 10:19, Stefan Schmidt a écrit :
François BERGANZ schrieb:
Hello all,
I need that Asterisk Always use Reason in a CANCEL.
How to do?
thank you
*François *
hello,
i
HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call
is always a ring group called '600', my problem is that after press 1 (but
this problem is present also with press 2) before that the inbound call is
transfer to extension pass 10/11 seconds !
In attach log file about
On Mon, 2010-05-24 at 15:09 +0200, Sasa wrote:
HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call
is always a ring group called '600', my problem is that after press 1 (but
this problem is present also with press 2) before that the inbound call is
transfer to
Hello everyone!
I haven't seen anything new about the JACK support in Asterisk and I was
wondering, if anyone has experience with a current release of Asterisk, JACK
and mISDN/googletalk etc. I'm thinking of installing a new version
(havingcurrently 1.60-beta9. But the excercise would be
Hello everyone,
I have asterisk installed in our call centre with aheeva platform and centos
linux,
We have 2 access provider I have configured the etc/asterisk/extensions.conf
in order to do the routing of calls
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten =
Hi all,
Any good TTS (free or commercial) for asterisk?
Rgds,
Ringo
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On Mon, 24 May 2010, Rilawich Ango wrote:
Any good TTS (free or commercial) for asterisk?
I like Cepstral with the Allison (Smith) font. Allison Smith does the
sounds distributed with Asterisk.
--
Thanks in advance,
-
Thanks. Do it support multi-language?
On Mon, May 24, 2010 at 11:55 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Mon, 24 May 2010, Rilawich Ango wrote:
Any good TTS (free or commercial) for asterisk?
I like Cepstral with the Allison (Smith) font. Allison Smith does the
sounds
I'm trying to solve a problem I have with agents hanging up on callers
before they even talk to them (caused by agents dropping their handset
or something.)
What I want is something like AgentLogin() where the agent has to press
'1' to accept the call. Does anyone know how to get this to work
On Mon, May 24, 2010 at 7:31 AM, Marcin J. Kowalczyk
marcin.kowalc...@ccig.pl wrote:
Medium load system (~300 simultaneous calls) crases few times a day.
1.6.1.19 but then upgraded to 1.6.2.7 but it's not solving issue.
Any idea what can be wrong/tunned?
I've three times had unexplained
The Cepstral paid version has several languages available and other voices
for those 10 people who don't like Allison. At $35.00 a pop, it's not
prohibitive (Lumenvox is much more pricey)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Found this at voip-info:
Not implemented SIP security in Asterisk
* *SIPS - SIP over SSL over TCP*. Since SIP over TCP is not
implemented yet (2003-09-10) - SSL over TCP is propably far away.
* *Update*: (Jan 2006) SIP over TCP is getting closer, see bug/patch
4903
Troll the archives over the last 6-9 months for VPN; some have tried this
with varying results.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, May 24, 2010 12:57 PM
To: Asterisk Users
On 05/24/2010 01:01 PM, Danny Nicholas wrote:
Troll the archives over the last 6-9 months for VPN; some have tried this
with varying results.
If you look for 'SIP TLS' instead of 'SIP SSL', you'll find a lot more
useful information, because SSL became TLS many years ago, and the SIP
RFCs only
Well, I don't want to take this too much off-topic, but intent is in
law an extremely difficult charge to prove. You basically need to have
witnesses confirm that the accused told them unambiguously that this is
what he had intended to do.
It is one of the most challenging prosecutions to
I am trying to get a zaptel install converted to dahdi.
I can get dahdi installed, and the pseudo device even shows up; however, dahdi
show channels shows me nothing. There is a TE122 and a TDM800 in there, and
neither show up.
dahdi show status shows both cards, and dahdi tools show that the
Hi,
I try to compile mISDN 1_1_9_2 (kernel: 2.6.32.13). I had the following
compiling error:
make[1]: Entering directory `/usr/src/linux-2.6.32.13'
CC [M] /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.o
/usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c: In
Michael Munger wrote:
dahdi show status shows both cards, and dahdi tools show that the cards are
there, working, and have no alarms.
What am I missing?
What are the contents of:
/etc/dahdi/system.conf
/etc/asterisk/chan_dahdi.conf
Doug
--
Ben Franklin quote:
Those who would
On Mon, 2010-05-24 at 16:56 -0400, Michael Munger wrote:
I am trying to get a zaptel install converted to dahdi.
I can get dahdi installed, and the pseudo device even shows up; however,
dahdi show channels shows me nothing. There is a TE122 and a TDM800 in there,
and neither show up.
You did a dahdi_cfg -vv to see that the channels are up?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, May 24, 2010 4:05 PM
To: Asterisk Users Mailing List - Non-Commercial
Hi!
I try to compile mISDN 1_1_9_2 (kernel: 2.6.32.13). I had the following
compiling error:
You need to use mISDN v2 with this new kernel. Be aware, mISDN v2 is a
better yet very different animal compared to v1. AFAIK chan_misdn only
works with mISDN v1, which means you will have to turn to
I am also very interested in support for Jack in recent release. I
envision a little project using Jack to route call audio into a digital
audio workstation (Neundo or Pro Tools) for real-time processing using
VST plug-ins.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
Thanks. Actually, I am looking for a TTS that support Chinese
(Mandarin and Cantonese). Do you have any suggestion? Up to now, I
can't find any TTS can support Chinese.
As I know Lumenvox is a voice recognition engine. Is it also a TTS?
ango
On Tue, May 25, 2010 at 12:51 AM, Danny Nicholas
On Mon, May 24, 2010 at 04:56:09PM -0400, Michael Munger wrote:
I am trying to get a zaptel install converted to dahdi.
I can get dahdi installed, and the pseudo device even shows up; however,
dahdi show channels shows me nothing. There is a TE122 and a TDM800 in there,
and neither show
Hi All,
I have the latest AsteriskNow installed (1.5) and after a couple of months
with system in production I have a need to use the Paging/Intercom features.
I have the module installed and I am able to successfully intercom with
individual phones using *80xxx (extension number) but if I create
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