search for: caescu

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