Displaying 19 results from an estimated 19 matches for "usedcanon".
2004 Jun 11
3
Background Playback fails
Hi Guys.
I've had a lay off from Asterisk for 12 months but I am starting to look
into it again. I am not very Linux savvy and found it hard going the
last time. I've started playing with it in the last 3 weeks and I have
to admit to making more head way this time.
The first problem I'm stuck on and I cant find a solution to is that
sound files that I have recorded (be it by
2004 May 27
4
AGI Pascal
Hi,
Has anyone done any AGI scripting in pascal. I would appreciate help anyone
can offer. My understandin on AGI scripting is very flaky, I am assuming
whatever language is used the application needs to be compile and made
executable. So if I write a script in pascal, I would compile it with
something like freepascal and make it executable.
Thanks
Umar Sear
2004 Aug 13
11
asterisk in india
Does anyone know if the E1 cards that digium sells work in India. Also are
there any distributers for those cards in India. By E1 cards I mean E100P,
TE410P or TE405P
--
regards
Vikram (http://www.vicramresearch.com)
2004 Jun 12
9
Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
How do you prepend. I want to be able to dial 7 digits instead of of
11 for local calls.
Can someone post there extensions.conf part that is relavent?
2004 Jun 12
5
MWI on Cisco ATA-186 (SIP)
I am trying to set up the Message Waiting Indicator (stutter
tone/light) so that my cisco ata-186 will let my phones know there is a
message waiting. However this does not seem to be very well
documented.
I found this on wiki mailboxnumber@context ... where does that go? Do
I put it in my SIP.conf definition for my cisco ata, or where. In my
SIP cisco definition i already have a
2004 Jun 11
3
ssh key problem
Hi I've need to reinstall my asterisk software (hard drive failure). I'm
back and running to a make samples state.
I have backed up all of my conf files (ok so they were about a week old
but much better than starting from scratch), the problem I am having is
with WS_FTP Pro.
Basically I used to connect to my asterisk server using this software no
problems just using root as
2004 Jul 04
1
How to use return value in extensions.conf
Hi,
I am trying to implement a dialplan in which the user is notified of a
missed call, if no voicemail is left.
Basically what I would like to achieve is something like ...
exten => _0207XXXXXXX,1,DIAL(SIP/${EXTEN},15)
exten => _0207XXXXXXX,2,HasNewVoicemail(${EXTEN:4}@default:INBOX|msgcount)
exten => _0207XXXXXXX,3,Voicemail(u${EXTEN:4})
exten =>
2004 Jul 28
2
Asterisk voicemail from mysql no longer working
Hi All,
I hope someone can help.
I have a system that I have recently upgraded to
latest CVS and my voicemail is not working from mysql
database.
I get an error on the console saying
" No entry in voicemail config file for 'number'"
whilst there is an entry in the database for the
specified number. It seems like app_voicemail is no
longer checking the database even though
2004 Sep 28
1
asterisks queues with static members
Hi List,
Forgive me if this has already been covered. I did go through past
messages but could not find anything.
I want to setup a queue like scenario where users don't need to
login/logout.
Basically I want to define a list of extensions that will be rung when a
call comes in. The sequence in which the extension are rung needs to be
intelligent, in the way queues are.
For example, it
2004 Jul 12
4
call Intrude
Hi
I have looked through the wiki and search the mailing list, but I cannot
find a way to intrude on a call, can asterisk do this feature?
if so how?
Thanks for your help
Robb
2004 Jun 10
3
Iax2 ringtone problem
Hi,
i have a problem with iax2 and ringtone.
Here is the call path
pstn -> asterisk -> iax -> firefly or any iax phone.
My problem is when i receive a call on my iax phone, the ring sound is very distort and bad.
If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal.
Otherwise, it is like a machine gun with iax
Help would be really
2004 Jun 10
1
RE: question about prepaid app_prepaid
...}
+
res = ast_bridge_call(chan, peer, allowredir_in,
allowredir_out, allowdisconnect);
if (res != AST_PBX_NO_HANGUP_PEER)
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of usedcanon
Sent: Thursday, June 10, 2004 5:08 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
I would be interested to share ideas, if you have guidence to offer I would
be greatful
Umar.
-----Original Message-----
From: asterisk-users-admin@lists...
2004 Jun 10
4
How to get the Called id with AGI
Hi all,
Is there a way to get the "called id" (the B number) with AGI perl ?
I know how to get the caller id which is working fine and is just below:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
$callerid = $input{'callerid'};
$AGI->say_digits($callerid);
}
Thanks in advance,
Angel.
2004 May 28
16
Asterisk Receptionist manager program.
We are writing a program using the manager for * for our receptionist
to use once the system go live. If anyone is interested in helping us
with testing please let me know.
We are designing it for a touch screen monitor for her to do transfers,
see whose on the phone and a few other features. Its in the development
stage and has bugs.
but I think its gonna be really good.
If your interested
2004 Jun 10
3
FW: question about prepaid app_prepaid
Hi,
I have compiled and installed app_prepaid module. But have problem when
connect to postgres database. I guess so because after key in card number,
it always play prepaid-no-aaa voice file.
Anyone succeeded in configuring the app_prepaid for prepaid calling service
for asterisk? Please help.
Ps: where can I view the log file for this module.
Thanks.
Tom
--------------
2004 Jun 11
11
Broadvoice and DTMF
I understand there has been some issues sending DTMF tone through
Broadvoice. Can some provide me with symptoms?
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004
2004 Jul 08
2
Shady dial anyone??
...list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--__--__--
Message: 5
Date: Thu, 8 Jul 2004 13:02:41 +0100 (BST)
From: =?iso-8859-1?q?Umar=20Sear?= <usedcanon@yahoo.co.uk>
To: asterisk users <asterisk-users@lists.digium.com>
Subject: [Asterisk-Users] Minimum install required for Asterisk + voicemail
& SIP friends from mysql
Reply-To: asterisk-users@lists.digium.com
I have been trying to install asterisk with MySQL for voicemail and SIP
frie...
2004 Jun 15
2
using SetCDRUserField in an AGI script
Hi I am trying to use SetCDRUserField in an agi script
but with no success.
I am using the CDR mysql addon, however I can't see it
being at fault as my attempt is not doing anything to
the CVS CD either.
has anyone used this, any hints guidence would be
greatly appreciated.
The syntax I am using is like so ..
res=DoExec('SetCDRUserField','12345');
and then dialing the