On 7/6/2011 4:36 AM, bilal ghayyad wrote:
Hi All;
I know that incoming calls for the agent can be recorded, but how I can let the
outbound calls for the agents to be recorded? I can determine the directory to
store the outbound calls of the agents to be other than the directory to store
the
On 3/31/2011 3:05 PM, Michelle Konzack wrote:
Hello Hans Witvliet,
Am 2011-03-31 22:24:50, hacktest Du folgendes herunter:
Hi Michelle,
Perhaps i'm not understanding your question correctly.
From what i read, i seems that you got your huawei working correctly as
an umts/hspa-modem, But now
James Lamanna wrote:
Hi,
Does anyone have any good empirical data suggesting what the maximum
number of PRI calls (incoming and outgoing)
without hardware echo cancellation can be handled on a single box is?
I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of
D-Channels
Phibee Network Operation Center wrote:
Hi
I have a problems with a new Asterisk Server,
when i want call, i have:
[Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160
handle_request_invite: Call from 'PHISIP01' to extension
'00420225352184' rejected because extension not found.
Tom O'Connor wrote:
On Tue, Jun 30, 2009 at 3:12 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com
mailto:tilgh...@mail.jeffandtilghman.com wrote:
On Tuesday 30 June 2009 08:24:29 Tom O'Connor wrote:
I'm currently
pointing fingers at either the hardware (someone on
I just installed 1.4.23.1 with the queue realtime logger backport. Here
are my configs:
musiconhold.conf
[default]
mode=files
directory=/var/lib/asterisk/moh-native
random=yes
queues.conf
[7703]
wrapuptime=0
timeout=15
strategy=rrmemory
retry=5
queue-youarenext=queue-youarenext
Tzafrir Cohen wrote:
On Tue, Jan 06, 2009 at 09:39:36PM -0600, Alejandro Kauffmann wrote:
Tzafrir Cohen wrote:
On Tue, Jan 06, 2009 at 02:28:53AM -0600, Alejandro Kauffmann wrote:
I've built SVN-trunk-r167180 and try to start it with:
asterisk -f -C /etc/asterisk/asterisk.conf
which
I've built SVN-trunk-r167180 and try to start it with:
asterisk -f -C /etc/asterisk/asterisk.conf
which results in:
Unable to open pid file '/var/run/asterisk.pid': Permission denied
Unable to bind socket to /var/run/asterisk.ctl: Permission denied
However, /etc/asterisk/asterisk.conf has:
Tzafrir Cohen wrote:
On Tue, Jan 06, 2009 at 02:28:53AM -0600, Alejandro Kauffmann wrote:
I've built SVN-trunk-r167180 and try to start it with:
asterisk -f -C /etc/asterisk/asterisk.conf
which results in:
Unable to open pid file '/var/run/asterisk.pid': Permission denied
Unable to bind
Of Alejandro
Kauffmann
Sent: den 5 december 2008 05:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] set monitor_filename
Ralf Träskman wrote:
Hi
I have this in my queue extension and I see this in asterisk when I call
to the queue
Ralf Träskman wrote:
Hi
I have this in my queue extension and I see this in asterisk when I call
to the queue, but no file is created in the directory any ideas?
exten =
s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID})
-- Executing [EMAIL
dubravko caric wrote:
Hi all,
I have a question regarding connection of two Asterisk servers to our
PBX. Each Asterisk server has one PCI E1 card, and they are in failover
mode with Linux HA. On our PBX we have only one E1 card towards Asterisk
servers.
My question is how to connect
Steve Anness wrote:
Thanks for the all the help, I have been pulling my hair out
I now have the trunk working in both directions. However, how do I add
voicemail capability?
exten = _11XXX,1,Dial(iax2/colo/${EXTEN:2},20,Ttr)
exten = _11XXX,n,Voicemail(${EXTEN:2:3}|su)
Thinking that
Steve Anness wrote:
I posted earlier in the day about needed help with IAX trunking. I did
some more reading and made some more changes.
Here is what I have thus far:
Iax.conf on one server
[general]
bindport = 4569
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
Carlos Chavez wrote:
I have a customer that wants to use meetme but they want to have the users
record their name so it is played to the other people on the conference. Is
there an easy way to do this?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A.
Tobias Ahlander wrote:
Date: Tue, 02 Sep 2008 18:08:52 +1200
From: Paul Crane [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Subject: Re: [asterisk-users] Asterisk Queue's
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Alexander Lopez wrote:
I think it would be a good idea to start an item in the Wiki about this.
Can anyone else chime in for their countries??
Others in the EU, Eastern, Far East?
So Far I have:
Australia:PSTN to PSTN and Cell to Cell are OK , but Cell to PSTN and
PSTN to Cell are
Mike Trest - Personal wrote:
Hi,
Can someone point me to a zapata.conf example that will create a
single DIAL OUT
group including all 4 spans on a TE4XXP?
One friend says to change the group number all to 1 on all 4 spans.
Another suggestions says it is possible to have these unique groups
RE Kushner List Account wrote:
Bill Hackensack wrote:
I realize my messages may seem rude and obnoxious, but let's face it,
I'm just saying what the rest of you are thinking. I learned by
reading, reading, and reading.
sarcasm
Nobody wants to search, nobody wants to read - HOW
Carlos Chavez wrote:
I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico.
This is really the first server I have used with PRI in Mexico as we
normally use MFC/R2. Everything seems to be working except that some
numbers always seem to be busy when you dial them. All these
David Kennedy wrote:
Hi
While I have fixed the problem from this post, I do have another
problem, and you have asked for a debug output here, so I'll go
against my better instinct and reply here :)
-- Making new call for cr 32774
-- Requested transfer capability: 0x00 - SPEECH
[
Jerry Geis wrote:
I have a box with a TE210P. Things work for a while then stop when
making call files.
I get NOANSWER as the return code (right away).
I am running asterisk 1.2.12.1, libpri 1.2.3 and zap 1.2.9.1
When I try to update to newer zaptel the machine locks when loading the
Doug Lytle wrote:
Tzafrir Cohen wrote:
On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote:
stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and
now swear (by?) 1.2 or 1.4.
My decision based on what I've been reading in the bug tracker and
people
So the questions: Is there anyway to further verify that
asterisk is
not sending any extra digits or filler digits to the telco on
the PRI? If the problem is not in asterisk or zaptel, what do
I say to the
Telco to get them to believe the problem is on their end?
At the console
Anyone have any idea if there is some sort of limitation to
the number of SIP or IAX end points which can register to an
Asterisk system (2.8Ghz dual processor, 2GB ram) while also
handling 30-50 simultaneous calls without getting into trouble?
Of course the 30-50 simultaneous calls end
That is interesting.. Not sure though how getting rid of IAX
could have fixed your SIP issues, seems odd.
We can't really get rid of IAX, our customers would flip their lids.
The big difference we have is that this has happened on more
than one occasion when there was little to no call
Whats the difference between the following statements in extensions.conf
include=inbound
AND
#include inbound/*.conf
The first one includes a context the second one includes a file(s).
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.432 / Virus
I see that the digium card doesn't share the IRQ however
Digium has recommended diabled USB still... additionally the
Digium card is on 169 which isn't a valid IRQ.. how can I
find out what it is sharing with?
lspci -vb will give you the irq as seen by the cards on the PCI bus
--
No
How do you guys manage that issue? Do you record a message (sorry, the
number dialed can't be completed) and play it when the PRI or BRI
returns a specific code? And what code is that?
We check HANGUPCAUSE and playback messages we recorded depending on value.
Check
Title: Message
Does anyone know if you can have multiple
TE110P cards in one chassis?
As
long as you make sure they don't share interrupts, sure. We have several
boxes running just fine with anywhere from 1 to 4
TE110P.
___
--Bandwidth and
Hello!
In my relentless quest for knowledge, I pose this question: who's got the
biggest dialplans, and how big are these monsters?
Our small contribution...
1175 extensions (2580 priorities) in 303 contexts
Alex
___
--Bandwidth and Colocation
suggestions. Where did you fetch your rpms?
I had to fix up the init scripts for everything to work
On 7/24/06, Alejandro Kauffmann [EMAIL PROTECTED] wrote:
My /etc/sysconfig/zaptel configuration has only one MODULES
directive
enabled MODULES=$MODULES wctdm
However when I start asterisk it loads
can you explain? I don't find any information on it... is this a tool or a
library?
Look at http://www.voip-info.org/wiki-Asterisk+standard+extensions
There's an example on how to setup hints under standard priorities.
___
--Bandwidth and Colocation
My /etc/sysconfig/zaptel configuration has only one MODULES directive
enabled MODULES=$MODULES wctdm
However when I start asterisk it loads the wct1xxp module. Which
configuration file controls the loading of card modules?
Check /etc/modprobe.conf I clear that out and just leave the module
on 1.2.4 and 1.2.7, we have to set the 'type=peer' for call-limits to
work effectively.
type=friend doesn't seem to enforce call limits at all.
if you haven't tried type=peer, try that first.
No, this doesn't work.
I believe you need to setup hints for call-limit to work.
Has anyone had any experience running asterisk on a dual-xeon HP Proliant
server. Have you had any experience setting up digium cards on this?
We have asterisk running on a DL140 dual-xeon (only 1 proc atm) with 1GB of
ram and a TE410P. Results are mixed. The bios has 3-4 options you can
RHEL 4 and therefore CentOS 4 had a bug introduced in the latest kernel.
https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568
This bug report has a typo as well. It should read:
#define DEFINE_RWLOCK(x) rwlock_t x = RW__LOCK_UNLOCKED
Fix the line and recompile zaptel. All should be
You need to run make from /usr/src/asterisk and not
/usr/src/asterisk/channels/h323. Just make then make install.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hussain Umair
Sent: Wednesday, January 04, 2006 5:35 PM
To: asterisk-users@lists.digium.com
Hey everyone here's my problem.
Have a queue configured, it plays the desired recording, checks to see if
agents are logged in via agentcallback, forwards the call according to
distribution method, times out according to timeout settings, logs out the
agent that did not answer, hunts for next
Atxfer is only available in HEAD not stable.
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Redstone
Sent: Tuesday, May 03, 2005 12:48 PM
To: Asterisk User
Subject: [Asterisk-Users] IAX2 attended transfer on 1-0-6 Stable
Hi Guys
I'm still
We installed AAH .05, tweaked it and learned more about dialplans, Queues
(not included in that version of AMP), upgrading to CVS Head (needed
atxfer/automon) and anything else we needed to scale AAH to our needs (75
agents 15k+ calls/day)than I believe we would have learned by simply trying
to
Of course I am not a kernel expert, so .. please be patient.
I am investigating on my zaptel/zapata problem.
As the main error message asterisk quits on mentions '/dev/zap/channel':
No such file or directory I went
peeking over there.
[Asterisk Verbose Error
Mar 13 20:43:35 WARNING[5779]:
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