Displaying 20 results from an estimated 5000 matches similar to: "PHP - Queues - etc."
2007 Aug 28
1
deadagi and billsec or answeredtime
Hello,
I want to create php rate script and I'm using Deadagi. But I allways get
billsec 0 , or nothing. Can you help me to solve this problem...
My extension.conf:
exten => _123,1,DeadAgi(rate.php)
exten => _123,2,hangup
And my simple test php script rate.php
#!/usr/local/bin/php -q
<?php
include_once (dirname(__FILE__)."/phpagi.php");
$AGI = new AGI();
2008 Oct 02
1
Asterisk Queue question
When the asterisk a queue reset their counters?
I 'm talking about this kind of info in asterisk console.
>show queue 600
600 has 0 calls (max unlimited) in 'ringall' strategy (4s
holdtime), W:0, C:14, A:8, SL:0.0% within 0s
I just say that because I have a queue with strategy "Fewest Calls"
working for a couple of mouths, and a new agent has been added this
2007 Oct 17
3
Play sound on hangup
Hi,
Does anybody have some ideas - how to play a sound file on channel, after that
bridged channel got hanged up?
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
2008 Feb 18
2
SiP call generator
I want to have a PC-based real-time VoIP bulk call generator (including both
SIP signaling and RTP generation)
for stress testing and precise analysis of the VoIP network equipment.
Do any one knows a free program can do that .
Regards
*********************************************
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with
2008 Nov 21
2
Log level of 500 Server Internal Error.
Hi,
VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error"
I just noticed that i sometimes get those back from provider. They are
currently general SIP message log entries with verbose level 3.
I wonder if such SIP fails could generate at least WARNING in log?
Currently i'm checking logs for warnings and errors, so i probably
have missed those.. It would be
2008 Dec 18
2
Asterisk 1.4.22 Queues problems (Fifo or not ?)
Hi,
I'm having a question with asterisk queue system, is it a fifo or a lifo
or random ?
Sometimes when we have people waiting in the queue and new agents are
connected to handle the load the first call that is handled is not the
one which
is already waiting for 4min, but the new one which has just arrived.
However this doesn't happens everytimes
Is it normal ?
regards,
benoit
2008 Nov 28
1
Priority between calls from different queues
Hi!
I want to know the way that calls are answer in this case...
I have queue1 and queue2, one agent that receive call from both queues.
queue1 <- call1
queue1 <- call2
queue2 <- call3
queue2 <- call4
In my test the agent answer calls in this order: call1,call3,call2 and
call4.
I think this must be in this order call1,call2, call3, call4 like a big
FIFO.
Its ok this behavior?
Could
2008 Nov 27
5
Any 1.6 SendFAX example ?
Hi,
Do you have any example showing how to use SendFAX ?
I can see several examples of ReceiveFAX but not a single one showing
SendFAX.
Regards
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2009 Apr 25
3
Outgoing Queues
Anyone thought about something like outgoing queues?
I mean, having same info that has for inbound queues but for outbound calls,
and grouping members there.
For example, before using dial application put an app outqueue that get all
the statics.
Talked time, member status, last call, completed calls, failed calls, reset
statics, and maybe some more. So its possible to get more control and has
2008 Oct 27
11
Fring: Open VPN client to be installed on the mobile, which mobile?
Hi All;
I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile:
Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server?
Regards
Bilal
2008 Sep 02
4
AgentCallbackLogin AddQueueMember
Hi
i have problem with AddQueueMember logic.
I need login Agent(Member) in asterisk.
use this option:
for example:
AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13)
and now i want to call to this Agent:
exten => _1XX,1,Dial(Agent/${EXTEN:1})
call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga.
This doesn't work, How can i do this on Asterisk 1.4(not
2007 Dec 25
1
Softphone to be installed on the Mobile
Thanks a lot for the help.
But if Asterisk has private IP address and the only
way to access it from remote sites is to have vpn
connection to the site that asterisk existed (the site
has vpn), then how that will happen from the Mobile to
be able to run the softphone from the mobile?
Any help?
Regards
Bilal
-----------------
I installed out of curiosity today, and guess what?
You can do SIP
over
2008 Mar 11
3
Call tracing - Asterisk 1.4
Hi guys
I've just read this about the upcoming release of * 1.6:
?Better reporting through a new call event logging capability in Asterisk
1.6 will allow complete tracking of events that take place during a call.
The goal, according to Fleming, is to provide more detail than traditional
CDR (Call Detail Recording) features offer and to allow for more granular
tracking and auditing.?
That
2008 Oct 10
3
Compile logger-mysql.c with UNDEFINED REF to `mysql_error'
Sorry to post a C compile error on this mailing list but this is
Asterisk related.
Basically, I was following
http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queu
e_logging
to patch logger.c and Makefile in Asterisk 1.4.* in order to write
queue_log to mySQL database.
When I ran make, it complained:
In function `write_mysql_logger':
[...]
2007 Oct 03
1
Resolving digit strings using pound/hash.
Hi all,
The thing that has bugged me about Asterisk since I first started
playing with it, is the fact that the pound sign/hash/octothorp doesn't
resolve digit conflicts or cancel timing on a variable length string such
as a tie line code or when you call numbers in a country whose length can
be different between numbers in the same plan. In North America, we see
this when calling
2008 Mar 17
6
Handling 3 different call ending causes
Hello List,
I'm using a dialstring like the one below. I want to have three different
things happening depending on exit cause.
Dial(SIP/${phonenumber},20,gL(20000[:5000][:5000]))
These 3 things could happen:
1, Caller hangs up
2, Callee hangs up
3, The 20 seconds is up and call is terminated from Asterisk.
Is there a way to separate these 3?
Thanks,
Best regards,
Tobias
--------------
2008 Nov 20
1
Macro conversion in 1.6
I create my sip users using a common macro in 1.4:
[internal]
exten => 200,1,Macro(phones|200|SIP/200)
[macro-phones]
exten => s,1,Dial(${ARG2}|45|Tt)
etc...
But now in 1.6 this fails:
-- Executing [200 at handsets:1] Macro("SIP/201-0942b530", "phones|200|SIP/200") in new stack
[Nov 20 08:55:55] WARNING[5958]: app_macro.c:201 _macro_exec: No such context
2008 Nov 23
1
Asterisk 1.6 mysql cdr log problem
Hi all!
I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers
and tools but my calls aren't logged. I'd enabled mysql log and
noticed that asterisk send a 'DESC cdr' so connection is working
between asterisk and mysql and I am able to call other phones so
Asterisk is working as well. No error messages on startup though.
Any idea why is it happen? As I realized
2007 Sep 21
1
Authenticate() application and CDR
Dear all,
I'm trying to configure Asterisk to be able to ask the caller to enter a
given password in order to continue dialplan execution. I've tested this
feature using the Authenticate application like this:
exten => _X./5219,1,Answer
exten => _X./5219,2,Authenticate(1234,a)
exten => _X./5219,3,Playback(pin-number-accepted)
exten => _X./5219,4,Dial(SIP/${EXTEN},120)
2007 Sep 24
1
# to transfer calls
Hi all,
I wonder why my call was transferred when I pressed '#' in a conversation.
How can I disable this kind of call transfer?
Thanks.
David
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