On Mon, Jun 6, 2011 at 9:42 AM, A E [Gmail] all.efor...@gmail.com wrote:
Hello,
using 1.8.4. using a very simple local AGI script in bash which has only
one line in it:
echo -e 'STREAM FILE welcome 123 \n'
dialplan:
exten = 5150,1,Answer()
same = n,Set(CHANNEL(language)=en_AU)
same
On Mon, 6 Jun 2011, A E [Gmail] wrote:
Hello,using 1.8.4. using a very simple local AGI script in bash which
has only one line in it:
echo -e 'STREAM FILE welcome 123 \n'
What gives? spent 2 hrs Googling but nothing! :(
Maybe 1.5 hrs should have been spent reading :)
One line does not an
On Mon, Jun 6, 2011 at 12:12 AM, A E [Gmail] all.efor...@gmail.com wrote:
Hello,
using 1.8.4. using a very simple local AGI script in bash which has only
one line in it:
echo -e 'STREAM FILE welcome 123 \n'
dialplan:
exten = 5150,1,Answer()
same = n,Set(CHANNEL(language)=en_AU)
same
On Mon, Jun 6, 2011 at 2:06 AM, mahesh katta maheshka...@flexydial.comwrote:
On Mon, Jun 6, 2011 at 9:42 AM, A E [Gmail] all.efor...@gmail.com wrote:
Hello,
using 1.8.4. using a very simple local AGI script in bash which has only
one line in it:
echo -e 'STREAM FILE welcome 123 \n'
On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, 6 Jun 2011, A E [Gmail] wrote:
Hello,using 1.8.4. using a very simple local AGI script in bash which has
only one line in it:
echo -e 'STREAM FILE welcome 123 \n'
What gives? spent 2 hrs Googling but
On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards
asterisk@sedwards.com wrote:
AGI is an interface. It consists of reading the AGI environment from
STDIN and then, writing requests on STDOUT and reading the response from
STDIN.
On Mon, 6 Jun 2011, A E [Gmail] wrote:
Right! I did read that,
On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards asterisk@sedwards.com
wrote:
AGI is an interface. It consists of reading the AGI environment from STDIN
and then, writing requests on STDOUT and reading the
On Thu, 2011-06-02 at 16:03 +0100, Ishfaq Malik wrote:
Hi Does anyone know of an accurate resource I could refer to for this?
The best I can find is
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
And that table wont create in my database...
Thanks
Ish
Can someone
Dear all,
I would appreciate it if you could teach me Asterisk SIP NAT Config.
I'm trying to capture SIP Register with externip that should set in
contact header at External SIP Server as shown below, but I haven't
seen it.
I need your help.
My experiment environment is as follows.
Hi,
Nobody on this?
Le 16/05/2011 23:35, Administrator TOOTAI a écrit :
Le 16/05/2011 18:27, Jose P. Espinal a écrit :
Administrator TOOTAI wrote:
Of course it's 1.4.41. And the result is that devices doesn't
register anymore.
Thanks for any hint.
If you are installing from source,
Hi
when I Upload MOH file from Asterisk GUI ,it is getting success and
even not getting any error,But if check the destination path the file is
not showing , even the source file and destination path and formate are
correct .I am not getting any error log from asterisk console too. I
read
Good Morning,
I'm thinking about buying the asterisk six-months online course,
Have somebody here that bought that course? What is your opinion?
Thanks.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards
asterisk@sedwards.com wrote:
I strongly suggest using an existing library for the language of your
choice.
On Mon, 6 Jun 2011, A E [Gmail] wrote:
Copy that. Not planning to write an AGI script in bash actually...it
will be written in C#
In article banlktikpxc_jk5xvhyx9akakltvl0v6...@mail.gmail.com,
A E [Gmail] all.efor...@gmail.com wrote:
Hello,
using 1.8.4. using a very simple local AGI script in bash which has only one
line in it:
echo -e 'STREAM FILE welcome 123 \n'
dialplan:
exten = 5150,1,Answer()
same =
I have a Bridged call with 2 parties. I want to redirect one party to a
conference room and the other party to an outside number. I tried doing that
with a dialplan. I used ChannelRedirect in the dialplan and redirected the
first channel to the conference room. however, the second channel
On 11-06-05 12:18 PM, satish patel wrote:
Hey guys!
I have just download latest SVN Revision 322051 and compile and install but my
asterisk -V showing still old version :( is it broken ?
/usr/sbin/asterisk -V
Asterisk SVN-branch-1.8-r321926
No, are still running the old binary? Also, did
Hello Folks;
Perhaps I am chasing my tail here.
Before I go any further, is this compatible/supported in Asterisk 1.6x?
If so, I would be willing to post any manager.conf or http.conf snippets
needed.
When I attempt to open the Asterisk Web GUI, I get a 'page not found'.
I am sure this is
On Mon, Jun 6, 2011 at 11:55 AM, Silver Thorne szilvertho...@gmail.com wrote:
Hello Folks;
Perhaps I am chasing my tail here.
Before I go any further, is this compatible/supported in Asterisk 1.6x? If
so, I would be willing to post any manager.conf or http.conf snippets
needed.
When I
On 06/06/2011 12:08 AM, Jeremy Kister wrote:
i'm trying to review issues that i'm monitoring and/or have reported at
http://issues.asterisk.org
when I click on 'My View' or 'View Issues' I get an error:
APPLICATION ERROR #401
Database query failed. Error received from database was #1142:
Hi,
I'm seeing some voicemails with skipping sound. Specifically, when somebody
is giving their phone number sometimes you can hear that a digit is missing,
but it was clearly said.it sounds like somebody just removed sloppily the
digit using a wav editor.
It might happen on live calls,
On Mon, Jun 6, 2011 at 10:39 AM, Tony Mountifield t...@mountifield.orgwrote:
In article banlktikpxc_jk5xvhyx9akakltvl0v6...@mail.gmail.com,
A E [Gmail] all.efor...@gmail.com wrote:
Hello,
using 1.8.4. using a very simple local AGI script in bash which has only
one
line in it:
echo -e
2011/6/6 Antonio Modesto mode...@isimples.com.br:
Good Morning,
I'm thinking about buying the asterisk six-months online course, Have
somebody here that bought that course? What is your opinion?
Thanks.
I have not bought the course nor will I.
I am self taught everything in IT and
Anyone have an update as to when Digium will ship a working package?
-- Forwarded message --
From: Andrew Joakimsen joakim...@gmail.com
Date: Wed, Mar 23, 2011 at 23:53
Subject: Issues with Digum Repos / AsteriskNOW Bad Packages
To: Asterisk Users Mailing List - Non-Commercial
On 06/06/2011 08:07 PM, Andrew Joakimsen wrote:
Anyone have an update as to when Digium will ship a working package?
According to https://issues.asterisk.org/view.php?id=18748 new packages
should already have been pushed. If not perhaps you could join #asterisk
or #asterisk-dev on
I have used those packages:
[Apr 7 01:09:51] WARNING[27966]: loader.c:434 load_dynamic_module:
Error loading module 'app_voicemail_imapstorage.so':
/usr/lib/asterisk/modules/app_voicemail_imapstorage.so: undefined
symbol: copy
[Apr 7 01:09:51] WARNING[27966]: loader.c:777
On Mon, Jun 6, 2011 at 10:10 AM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards asterisk@sedwards.com
wrote:
I strongly suggest using an existing library for the language of your
choice.
On Mon, 6 Jun 2011, A E [Gmail] wrote:
Copy that.
Hi all,
I have just upgrade asterisk 1.2 to 1.8 and we have numbers starting from
_71XX. now what happen if i dial any 711X number my polycom just dial 711 and
say busy number look like my phone doing some regex itself. like 911 number..
Did you get what i am trying to say ? it was working
look like we found issue in phone configuration files [2-9]xx
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Mon, 6 Jun 2011 20:43:22 +
Subject: [asterisk-users] asterisk 1.8 issue with polycom dialplan
Hi all,
I have just upgrade asterisk 1.2 to 1.8 and we
Hi all,
I've got something strange, that got me searching for quite awhile.
Configuration as followed:
Linphone on a laptop, that is connected via openvpn to a proxy.
That proxy is connected with iax to another asterisk.
On the second one i have several hard and softphones.
Behaviour at first
Way to go Steve. That's the best way to learn.
Steve Totaro stot...@asteriskhelpdesk.com wrote:
2011/6/6 Antonio Modesto mode...@isimples.com.br:
Good Morning,
I'm thinking about buying the asterisk six-months online course, Have
somebody here that bought that course? What is your
+1...I am an autodidact myself, never took any courses in IT or Telephony other
a computing course in tge late 80s that was actually a typing class that used
computers.
Slainte,
Sherwood McGowan
Sent from my iPhone
On Jun 6, 2011, at 5:58 PM, Amadu alsta...@gmail.com wrote:
Way to go Steve.
sometime i am getting Span 1: Channel 0/23 got hangup request, cause 16 but my
call doesn't get completed
== Primary D-Channel on span 1 up
-- Restart requested on entire span 1
== Using SIP RTP CoS mark 5
-- Executing [7076941815@from-sip:1] Dial(SIP/7328-0004,
Hi all,
We recently decided to get a professionally recorded set of prompts for
our asterisk based IVRs and received these as the following:
Bit Rate: 1536Kbps
Sample Size: 16bit
Channels: Stereo
Sample Rate: 48kHz
Format: PCM
I use Wavepad to convert it to:
Bit Rate:64Kbps
Sample Size: 8bit
This is wired..
If i connect my old asterisk 1.2 box my PRI working great! all inbound outbound
calls.. But its not working with asterisk 1.8 :( ( i can call in but not out)
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 7 Jun 2011 02:11:28 +
Subject: Re:
Dear
I installed chan-sccp-b v3 on a powerful virtual machine, with 4 cpu cores
and 16GB RAM(enabled in kernel by PAE)
about 1,200+ clients are going to register in this machine. all data of
clients are saved in ORACLE. The asterisk (1.6.2.18) connected to the
database throw odbc(unixodbc).
all
Solution:
pridialplan=unknow
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 7 Jun 2011 02:33:44 +
Subject: Re: [asterisk-users] PRI issue its BUSY
This is wired..
If i connect my old asterisk 1.2 box my PRI working great! all inbound outbound
calls..
Hi,
I had a similar issue converting wav files one time. Ended up using sox to
convert to .sln as that ended up being the sounding conversion.
I used the below command on a directory of files to convert:
for a in *.wav; do sox $a -t raw -r 8000 -s -w -c 1 `echo $a|sed
s/.wav/.sln/` resample
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj
Chand Sent: Monday, June 06, 2011 7:12 PM
We recently decided to get a professionally recorded set of prompts for
our asterisk based IVRs and received these as the following:
[snip]
The problem I have
On Jun 6, 2011, at 10:51 PM, Steve Edwards asterisk@sedwards.com wrote:
Sox has a bunch of obtuse (IMNSHO) commands. There may be one that could
automagically trim the pop for you.
The argument is question is the trim command. If the OP wishes to find an
automagic method, they would
On Jun 6, 2011, at 10:51 PM, Steve Edwards asterisk@sedwards.com
wrote:
Sox has a bunch of obtuse (IMNSHO) commands. There may be one that
could automagically trim the pop for you.
On Mon, 6 Jun 2011, Sherwood McGowan wrote:
The argument is question is the trim command. If the OP wishes
I use following for MySQL...
CREATE TABLE queue_log(
id int(11) NOT NULL auto_increment,
time datetime not null,
queuename VARCHAR(50),
agent VARCHAR(50),
callid varchar(32),
event VARCHAR(100),
data1 VARCHAR(100),
data2 VARCHAR(100),
data3 VARCHAR(100),
data4 VARCHAR(100),
data5 VARCHAR(100),
Hi List,
I am trying to get DTMF into conference room. for conference I am using
Konference module. Konference don't have an option of DTMF gets. Is there
any way by which I can get DTMF within conference room?
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
--
Hi List,
I am facing an issue of automatic DTMF created by Asterisk(1.4,1.6,1.8).
Issue is that when conference goes more then 10 minutes then we gets more
DTMF which is generated by asterisk.
The reason of starting these DTMF is loud volume, more noise area, Baby
voice and lady voice.
It's
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