Displaying 5 results from an estimated 5 matches for "caescu".
2009 Jan 16
0
No subject
...ssing.
If you want your application to initiate a call out without being
started through the dialplan:
* Asterisk auto-dial out Move (not copy) a file into an Asterisk
spool directory and a call will be placed
* Asterisk Manager API Use the Originate command
Regards,
Chris
2009/5/13 Dan Caescu <dcaescu at eqnet.us>:
> Forget the typo (s/ANSWERED/ANSWER/g)
>
>
>
> ________________________________
>
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Caescu
> Sent: Tuesday, May 12, 2009 7...
2009 Jan 16
0
No subject
...ssing.
If you want your application to initiate a call out without being
started through the dialplan:
* Asterisk auto-dial out Move (not copy) a file into an Asterisk
spool directory and a call will be placed
* Asterisk Manager API Use the Originate command
Regards,
Chris
2009/5/13 Dan Caescu <dcaescu at eqnet.us>:
> Forget the typo (s/ANSWERED/ANSWER/g)
>
>
>
> ________________________________
>
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan =
Caescu
> Sent: Tuesday, May 12, 2009...
2009 May 12
1
enum agi interesting problem
Hi,
I am having a strange problem with enum and AGI.
Here is what happens:
I have in my agi something like that:
foreach my $resolver ("e164.arpa", "e164.info", "e164.org") {
my @enums = get_enums($phone, $resolver);
foreach my $enum (@enums) {
$dialstring = $enum .
2000 Dec 10
7
load balance/redundancy
I have looked through the archives, and I can''t find the answer(not that
it isn''t there)
I have two connections to the net. I want load balancing and redundancy.
cable adsl
(24.141.) (64.229)
| |
| |
| |
| |
------------
| linux |
|redhat 7.0|
I have no idea were to even start. I would like equal access to both
connections.
2009 May 06
0
astcc - outgoing call does not hangup properly
Hi,
I am using ASTCC and trying to setup a calling card platform.
The problem that I have is that astcc does not hangup calls correctly:
1. If I try to dial a number, call goes through fine. When I hang up
the call from my side I get this:
-- Called 192.168.1.56/1XX6872XXXX (masked a few digits)
-- SIP/192.168.1.56-086c5000 is making progress passing it to