sorry test.
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Hi Mark.
Thanks for release this application!
It's works well !! great !!!
mack
On Wed, 14 Jul 2004 00:24:08 -0500 (CDT)
Mark Spencer [EMAIL PROTECTED] wrote:
I am hereby announcing the immediate availablity of ASTCC for *alpha*
testing. ASTCC is an AGI script and CGI script which greatly
no,
that mean use with gk.because that isn't bridge by *.
Does anyone knows another way can do that?
mack_jpn
On Thu, 24 Jun 2004 23:10:43 +0900 (JST)
Isamar Maia [EMAIL PROTECTED] wrote:
Nakano San,
Have you tried to make * only to route the connection and
they just talk
Hi all
How can I fix this problem?
Regards,
mack_jpn
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I tryed it.
but callee cannot answering with video in SIP.
# surely videosupport=yes in sip.conf
H.323 is works well but I think stilln't support over * yet.
mack_jpn.
On Thu, 24 Jun 2004 14:03:10 +0200
Michael Devenijn [EMAIL PROTECTED] wrote:
I found this tool, but didn't have the time
Hi all
I was tryed to connect to mysip.ch scs_client by siemens that isn't
works well.
Does anyones knows to work H/W or S/W applictations in asterisk SIP
videosupport?
Regards
mack_jpn
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Hi
could you please tell me your enviroment about this problem?
and how do you fix it ?
please disclose, when are you okay.
Regards
mack_jpn
On Fri, 2 Apr 2004 12:40:52 +0900
岩田 伸介 [EMAIL PROTECTED] wrote:
Hi.
At last,
Is this same result?
http://lists.digium.com/pipermail/asterisk-users/2003-November/027321.html
mack_jpn
On Thu, 1 Apr 2004 01:08:44 +0900
岩田 伸介 [EMAIL PROTECTED] wrote:
hi.
I got these compile errors while install asterisk.
readline
Hi Steve.
On Thu, 18 Mar 2004 22:06:46 +0800
Steve Underwood [EMAIL PROTECTED] wrote:
snip
There is now a new tarball at
ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1b.tar.gz Please try
this, and report any problems you find. This version has the following
changes:
A
That is fahdtel's product?
http://www.dairiten.com/modules/mylinks/visit.php?lid=217
It looks very cool.
mack_jpn
On Mon, 15 Mar 2004 06:23:22 +
Miguel Cavazos [EMAIL PROTECTED] wrote:
thanx for the review michael, could you send some pictures of the phone?
can you tell how long does
On 16 Mar 2004 01:08:27 +
Hermann Wecke [EMAIL PROTECTED] wrote:
On Mon, 15 Mar 2004, Matthew Marlowe wrote:
I can confirm 1.0.4.53 is bad as well. :) 1.0.4.50 has been working
fine for me.
I received the 1.0.4.54 firmware. So far, so good. No new problems.
How can I get this firm?
Hi all
I was tryed to use asterisk with alsa following versions.
alsa-driver-1.0.2c
alsa-lib-1.0.3rc1
but an error occured about that.
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 1074440928 (LWP 2141)]
0x41114e03 in snd_pcm_hw_param_set_near (pcm=0x80e9578,
yes I know two Web based iax client.
the one is
Babar Shafiq's iaxclient.ocx
http://www.geocities.com/babarnazmi/index2.htm
and another one is
Dan's activediax
http://www.laser.com/dante/diax/diax.html
so usefull :-)
We can find topics at here.
http://iaxclient.sourceforge.net/
mack_jpn
On Tue, 13 Jan 2004 16:55:03 -0500 (EST)
Greg Boehnlein [EMAIL PROTECTED] wrote:
I was toying with the idea of making an Knapterisk distribution based
on Knoppix. I've got my hands full right now, however, and can't really
get involved in another project.
Does anyone try it?
On Thu, 8 Jan 2004 21:25:55 -0500
Steven E. Frazier [EMAIL PROTECTED] wrote:
Is there an easy way to get all of your sound files, do you have ftp or just
http?
Thanks.
Steve
I think WebDAV is easy way. but I stilln't try it.
___
Hi Mike
I know exacty same situation about BT100 that sometimes lost any packets.
like a DoS attack for BT100? ;-(
mack_jpn
[EMAIL PROTECTED] asterisk]# ping 192.168.XX.XX
PING 192.168.XX.XX (192.168.XX.XX) from 192.168.XX.X : 56(84) bytes of
data.
64 bytes from 192.168.XX.XX: icmp_seq=0
On Thu, 01 Jan 2004 17:50:32 +0100
Philipp von Klitzing [EMAIL PROTECTED] wrote:
Hi!
We needed the client browser to be open all the time for dynamic data to
load without the page refreshing. After looking at all of our options we
decided on programming it ourselves using flash
.
snip
Apache + Mysql + PHP ( Ming + Actionscript ) + Asterisk is good.
Dynamic effective,Easy coding and Fast response :-)
---
Masakazu Nakano.
Dairiten.com - an open source VoIP and Ubiquitous Portal site in Japan.
http://www.dairiten.com/modules/news/
powered by xoops at http://www.xoops.org
but that's web admin page sometimes no respond and some hang-ups ;-P
I think that still dangerious for bussiness use.
mack_jpn
On Wed, 24 Dec 2003 08:08:13 +0200
Lubomir Christov [EMAIL PROTECTED] wrote:
I have to say that Grandstream phones are REALLY
I need more instruction too.
like a http://www.dairiten.com/webiax/
On Mon, 22 Dec 2003 00:02:08 -0200
Carlos Arnt [EMAIL PROTECTED] wrote:
I put like the readme.txt say the code in my web page , put the OCX in the same
Directory, but not work.
Did has any problem ??
Thanks
The code:
Hi sergio
On Fri, 19 Dec 2003 14:49:15 +0100
Sergio Serrano Revuelto [EMAIL PROTECTED] wrote:
Hi all,
I have tested RxFAX application through X100P card. When Fax
arrive i obtain the next trace:
snip
5 (0.01679,-0.16590) - 0.02781
6 ( -0.04451,
Hi all
I was tryed to use rxfax compiled with no problem.
and I get the FAX over pstn.but it only first 3 bytes.
like that.
-
II*
-
but not enough more.
How can I fix this problem?
and
I found two files of tiffiop.h and tif_dir.h at /tiff-v3.6.0/libtiff
Is that right? or
like that but poor :-)
// CDR access
$connect = @mysql_connect (localhost,USERNAME,PASSWORD);
mysql_select_db(asteriskcdrdb);
$result = mysql_query (select dstchannel from cdr where dstchannel like
'%IAX%',$connect);
$iax = mysql_num_rows($result);
$result = mysql_query (select dstchannel
Hi Isamar
maybe I think disclose your code to CVS is best and fast :-)
mack
Hi folks,
I'm trying now to play with fsk_modem.c and callerid.c
to get the Japanese callerid working and I already got to make some
steps..
I don't know if anybody accomplished that already... but
Since two
Hi Hachison.
you need voice actor and sampler software and codec conversions.
mack_jpn
On Fri, 21 Nov 2003 19:00:07 +0900
Hachison [EMAIL PROTECTED] wrote:
Hello
I can cange busy and unavaile message by voicemailmain.
I would like to change the Voicemail and Voicemailmain prompt itself.
On Tue, 11 Nov 2003 09:30:41 +0200
Dan [EMAIL PROTECTED] wrote:
Hi,
- Original Message -
From: Masakazu Nakano [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 11, 2003 2:18 AM
Subject: Re: [Asterisk-Users] DIAX version 0.9.2 available for download
On Mon
Hi Dan
Could you please compile with MBCS ( multibyte charactor supports ) okay?
http://msdn.microsoft.com/library/default.asp?url=/library/en-us/vccore/html/_core_international_enabling.asp
mack
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On Mon, 10 Nov 2003 17:40:02 +0200
Dan [EMAIL PROTECTED] wrote:
and need 'callto:' support :-)
Why you need this?
Give me an example.
1) netmeeting and cuseeme are already support.
2) make a call easier with groupware and/or portal.
( 'tel:' is good too )
mack
thanks Dan!
and I found an error. please fix.
strange function in call,when it stilln't registered
( like a no home server... ).
On Sun, 9 Nov 2003 19:53:36 +0200
Dan [EMAIL PROTECTED] wrote:
Hi all,
As promise, the new prerelease (0.9.2) is now available for download from
the
Hi Dan.
thanks for good application!
and I wish 'no with installer' package about that.
because I think use with USB-memory device in any places (ie.net-cafe.)
is that need registry setting or not?
On Sun, 2 Nov 2003 22:21:09 +0200
Dan [EMAIL PROTECTED] wrote:
Hi all,
I have developed a
I found it. but that webite is chinese BIG-5. take care.
http://www.mpn.com.tw/index-big5-PRODUCT.html
and that already released by Fujitsu.
http://www.net-2com.com/jp/product/hw/wireless_ipphone/
mack_jpn
On Fri, 03 Oct 2003 19:03:10 -0500
Steven Critchfield [EMAIL PROTECTED] wrote:
On
and I think use those cvs with rh7.3 and apt for RH is works well :-)
mack_jpn
Tilghman Lesher wrote:
On Monday 22 September 2003 22:37, Steve wrote:
On Monday 22 September 2003 11:25 pm, Andrew Joakimsen wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I bote 'with PHP' too :-)
it's so easy to access text strings,web,db,sounds(ecasound) and agi :-)
---
Masakazu Nakano.
Dairiten.com - an open source VoIP and Ubiquitous Portal site in Japan.
http://www.dairiten.com/modules/news/
powered by xoops at http://www.xoops.org
On Thu, 26 Jun 2003 21
Hi
We can found a couple of ITSP at Jasomi networks's PR.
http://www.jasomi.com/pr_deployment.html
Does anyone try it?
mack
On Thu, 12 Jun 2003 19:48:43 +0200 (MEST)
Martin Dommermuth [EMAIL PROTECTED] wrote:
Hi,
* Erik Lagerway
check here
http://www.metadecks.org/software/sweep/
---
Masakazu Nakano as [EMAIL PROTECTED]
On Tue, 18 Mar 2003 17:35:39 +
WipeOut . [EMAIL PROTECTED] wrote:
Hi,
What utilities do you guys use to record your IVR messages??
I know there is the record application in * but I was wondering
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