Displaying 20 results from an estimated 1200 matches similar to: "S(x) - Hang up the call after 'x' seconds - Not working from queue"
2004 Jun 24
2
How to force G729
We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729.
The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that:
[mypstngate]
type=friend
2006 Feb 08
1
odd 'digital' sound artifacts
Hi,
I've got some weird sound artifacts happening during calls, they're very
hard to describe, so I have a 122kb recording:
http://openprojects.rarcoa.com/~miztic/artifact.wav
normally the artifacts are just short blips, not quite as long as the
one above, but they sound the same.
When using the aggressive echo suppressor, it seems like those artifacts
cause a really loud buzzing sound to
2004 Nov 30
1
Agents/Queues - Drops call after 60 seconds
This just started happening today. I've got 1 queue and 6 agents. All logged
in. I tell the service people to ignore my call if they see my caller id.
I call the queue and watch as asterisk bounces me around the phones. Our
agent ring time is 5 second timeout and a 5 second wait time before trying
next agent.
I get the same message in console for each agent attempt:
-- Executing
2006 Apr 05
2
What causes deadlock?
Hi
What causes deadlock?
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x82acb10', 10 retries!
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x8298160', 10 retries!
Here is the portion of the log:
Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)...
Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing
2005 Aug 01
2
TDM400P REV I issues - ProSLIC vs TDM400P
The REV I card shows up in the PCI table as:
02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or
02:05.0 Class 0280: e159:0001)
Subsystem: Unknown device b119:0001
But the REV E/F shows up as:
02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface (or
02:0d.0 Class 0780: e159:0001)
Subsystem: Unknown device b100:0003
One
2005 May 18
1
Agent Queues and Sending URLs
Hi guys,
I'm testing the sending of a URL to an XLite softphone when a call is
in queue. See the output of the CLI below:
-- Executing Queue("Zap/69-1", "q_sample|tT|http://
www.google.com/") in new stack
-- Started music on hold, class 'default', on Zap/69-1
-- outgoing agentcall, to agent '1000', on 'Local/
1000@agents-1b94,1'
2005 Jun 17
1
callqueues confused :(
> -- Started music on hold, class 'default', on
> SIP/193.111.200.67-0815c790
> -- outgoing agentcall, to agent '1001', on 'Local/201@sip-0add,1'
> -- Called Agent/1001
> -- Executing Dial("Local/201@sip-0add,2", "SIP/101|20|tr") in new
> stack
> -- Called 101
> -- Agent/1001 is ringing
> --
2005 Sep 21
1
Addendum to Problem with Queues question
Here is the full "transaction"
-- outgoing agentcall, to agent '1001', on
'Local/3044@local-4fee,1'
-- Called Agent/1001
-- Executing Macro("Local/3044@local-4fee,2",
"sipline|3044") in new stack
-- Executing Dial("Local/3044@local-4fee,2",
"SIP/3044|20|t") in new stack
-- Called 3044
-- SIP/3044-ea92 is
2005 Sep 21
2
Problem with Queues
I am getting this on the console once people call in
-- outgoing agentcall, to agent '1001', on
'Local/3044@local-fd6d,1'
-- Called Agent/1001
-- Executing Macro("Local/3044@local-fd6d,2",
"sipline|3044") in new stack
-- Executing Dial("Local/3044@local-fd6d,2",
"SIP/3044|20|t") in new stack
-- Called 3044
--
2006 Jan 29
1
file.c:509 ast_openstream_full: File 100 does not exist in any format
Hi all,
look at these lines.
I created a queue named info when a caller (extension
86) place a call he is put on queue he sould hear MOH
.
What's the meaning of :
Jan 29 14:35:30 WARNING[2591]: file.c:509
ast_openstream_full: File 100 does not exist in any
format
Jan 29 14:35:30 WARNING[2591]: file.c:821
ast_streamfile: Unable to open 100 (format ulaw): No
such file or directory
Regards
2005 Oct 03
0
TDM400P recognised as "Network controller: Unknowndevice"
All the 'unknown device' means is that your 'lspci' doesn't know what
the card is. That's all. Nothing more.
--Rob
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aryanto
Rachmad
Sent: Tuesday, 4 October 2005 7:43 AM
To: asterisk-users@lists.digium.com
Subject:
2006 Jan 19
1
TDM400P zttest not working
Hi,
I have a TDM400P running with only one FXO port running on a VIA
EPIA MS10000 (1Ghz Via Eden). When I run zttest from Zaptel 1.2.1 it
hang and when I interrupt it with Ctrl-C that is the result: ?anyone
have some idea about why isn't working?
Some additional info:
# /usr/src/zaptel/zttest -v
Opened pseudo zap interface, measuring accuracy...
--- Results after 0 passes ---
Best:
2005 Oct 03
2
TDM400P recognised as "Network controller: Unknown device"
Hello everybody,
I have been googling for hours and also searched on http://www.voip-info.org/wiki-Asterisk, but I still can not find any information for the problem I have. So I hope one of you could help me out.
I have actually very little experience in Asterisk and also Linux. But by following installation guide, luckily I could get asterisk working. That is only with SIP and IAX channels
2005 Jul 23
2
(cause 66 - Channel not implemented) -- IAX?
Hi,
I am setting up a small call center using *. I have ZAP setup for
incoming calls and IAX setup for agents. Agents login using
AgentCallbackLogin. When customers call, it's getting picked up and when
queue is trying to call back the agents, I am getting error.
I am using CVS HEAD, and updated just now.
The error is:
-- Executing Answer("Zap/1-1", "") in new
2004 Dec 06
1
Queue Timeout
I am using the CVS version as of July 23, 2004. Is the queuetimeout
option not available in that cvs? If not, how do I go about applying
one of the queue patches without taking my system down?
2005 Sep 21
1
Call getting disconnected in queue
Hi,
I have a small call center with 4 Zap lines and 4 agents. Agents login
using sip phones with AgentCallbackLogin. I occasionally gets a
complaint that when customers call the call center, after the initial
greeting is over the call gets cut after playing the thank you message.
I started investigating and found that that happens when the call gets
transferred to an agent who is making an
2006 Oct 25
3
Maximum talktime in a queue?
Hi,
Is it possible to define maximum talk time in a queue? ie any one who
joins a queue should not be able to talk more than say 5 minutes to
the agent.
raj
2009 Oct 26
1
DAHDI not detecting RINGING Status on the Channel
I am using an 8 port tdm card and also I implemented a dialer using a
.call file generator. As you know on the .call you specify the channel to
call and then the contex/extension/priority to let dial plan continue when
the call is bridge.
My actual problem is that when the call process starts, asterisk (DAHDI)
sets the channel as answered when the truth is that on the other side the
channel has
2006 Apr 13
1
AgentCalled event
Hi,
I'm writing a Java client/server application that talks to the Asterisk
manager interface via the asterisk-java stuff. The idea being it will
give you an app to run on your desktop that monitors your phone
essentially. Once I've got something vaguely working it will be released
under the GPL and hopefully people will contribute to it etc...
As part of this, I'm currently
2007 May 21
1
CAS signalling conflicts with Clear channel
Hi,
My asterisk server was working with a 4-FXO analog
card (TDM400P).
I recently added two digital cards: a TE120P (1 PRI)
and a B410P (4 BRI).
The B410P is still unconfigured but inserted in a PCI
slot.
The TE120P's jumper is set to E1 as it will connect to
a commercial PBX's PRI card also configured as E1.
My analog channels used to be 1-4 but since I added
the new cards I changed