Displaying 20 results from an estimated 3000 matches similar to: "Odd Read App Issues"
2009 Mar 04
2
Bounty- CDR Bug Fix
I saw some of the heat about the $20 bounty earlier. So I don't want to
put a low bounty out.
Quote me a bounty, and I'll see if I can get it approved by management. :-)
I'm in need of getting this bug fixed. Bug has all of the details, but
basically 1.4.22 broke it all.
I've waited as long as I can - hoping the bug would 'resolve itself' -
but now I'm putting a
2009 Feb 23
2
Voicemail and ADSI
Anyone using this feature of asterisk's voicemail? I'd never heard of
ADSI, and saw it as I was perusing the voicemail source this morning. Is
it some kind of visual way of managing voicemail on your phone's display,
or does it require a terminal of some kind?
Thanks!
j
2009 Feb 20
2
zaptel telephone cards and asterisk in another pc
Hello,
I have some zaptel cards, and I would like to install them in some
user's computers. Is there any way to connect those cards with
asterisk server (which is in another computer)?
All manuals I have read explain how to connect asterisk and zaptel
cards in the same computers, but not on different ones.
Thank you very much.
2009 Feb 24
2
what is the correct character to separate application parameters: , or |
Hi!
I see lots of examples using , but core show application displays |
So what is the correct character to use to separate parameters for
application, functions and macros?
thanks
klaus
2009 Mar 02
5
How to generate core dump?
Asterisk segfaulted on me the other day; how do I tell it to generate a
core file so -- if it happens again -- I can attempt to debug? I looked
in the obvious places in "make menuconfig" and didn't see anything
appropriate.
Thanks,
-Ken
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
2009 Mar 31
0
Dead Call But Still Active
I'm having a strange issue, and not really sure where to even begin to
troubleshoot it. First let me explain that I have all agents setup
locally ( local/100 at agents/n)
A call will come in and ring to the agent. When the agent answers the
call, they just hear a dial tone. Agent hangs up. Asterisk still shows
the agent as 'in use' in queue status. And 'show channels'
2009 Feb 25
4
TE121 on Asterisk
Hello, I just bought a TE121 T1/E1 card, and now trying to install it on a
1.4.23.1 asterisk with dahdi 2.1.0.4
Actually first everything went on well and i managed to see my card on dahdi.
Here's the output:
#asterisk# dahdi_hardware
pci:0000:04:08.0 wcte12xp+ d161:8000 Wildcard TE121
and this is the scan:
--------------------------
asterisk# dahdi_scan
[1]
active=yes
alarms=RED
2009 Feb 25
5
AGI problem using mono (.Net)
Hello.
I have a software developer creating a .Net / mono program to use as an
AGI script. We are having problems getting it to stream files. From what
we can tell, it is talking to asterisk correctly when called from the
dial plan. Its stderr output goes to the asterisk console. But asterisk
doesn't give any indication that it receives the STREAM FILE command.
Asterisk simply quickly
2009 Jan 02
4
Setting Periodic-Announce filename in the dialplan
I'm wondering if there's a way to set which periodic-announce file is
played from my dialplan, much like setting the monitor-filename.
Something like this:
exten => s,n, Set(PERIODIC_ANNOUNCE=foo)
This would be a great feature if it doesn't already exist. Or perhaps
there's a better way to do this.
Thanks for your time.
--
Regards,
Robert Broyles
2003 Nov 16
1
na.rm default
Is there a way to set "na.rm=TRUE" as a default, so that this does not have
to be re-specified for all of the functions requiring this option?
Thanks in advance,
Stephanie Broyles
sbroyl@lsuhsc.edu
[[alternative HTML version deleted]]
2009 Feb 27
9
call file concurrency
Is there a convenient way to limit the number of call files (outgoing
directory) that are processed concurrently?
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/x-pkcs7-signature
Size: 3234 bytes
Desc: S/MIME Cryptographic Signature
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/a46e68fa/attachment.bin
2009 Mar 19
1
Overriding Queue Wrapup Time
Is there a way to override the queue wrapup time on the fly?
I would like to allow a longer wrapup time for my agents, but if they
are already done with closing up the call ticket, I would like them to
be able to dial an extension or something to override the wrapup.
Is there a way to do that?
--
Regards,
Robert Broyles
2015 May 01
3
DPMA - Asterisk Realtime
We love our Digium phones and DPMA - but we really need it to work on
our Realtime Platform. Otherwise we lose all the cool features and they
are just standard SIP phones.
Anyone working on a solution for this? Or anyone from Digium see this on
the roadmap?
2015 May 07
0
DPMA - Asterisk Realtime
On Fri, May 1, 2015 at 10:43 AM, Robert Broyles <robert at webservicesaz.com>
wrote:
> We love our Digium phones and DPMA - but we really need it to work on our
> Realtime Platform. Otherwise we lose all the cool features and they are
> just standard SIP phones.
>
> Anyone working on a solution for this? Or anyone from Digium see this on
> the roadmap?
>
Hey Robert -
2010 May 11
1
iax calls via checkbox.cc
Is anybody using checkbox.cc to make iax2 calls?
They have recently did some changes and my calls no no longer go through.
They don't have a best service either, not replying to emails..
--
Joseph
2009 Jan 05
1
CDR - What Changed?
On 12/17/08 I updated to 1.4.22 from 1.4.21...
Now the CDR data isn't recording calls where the caller hung up while
waiting on the Queue.
Sample CDR data BEFORE the upgrade:
"2008-10-30 12:46:47";"\"John\"
2009 Feb 06
1
AgentCallBackLogin no longer works after installing asterisk 1.6
Hi,
My queue used to work fine until I upgraded to 1.6. I am getting the
message:
No application 'AgentCallBackLogin' for extension (default, 31001, 1)
After some rearch I learnt that AgentCallBackLogin is removed in 1.6.
Any one has a configuration that works in place of AgentCallBackLogin in
1.6.
--
ond
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2009 Feb 18
1
trunk to trunk
Hi,
Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk of
Asterisk B (registered in Asterisk A as extension)
to incoming call across another trunk of Asterisk B to extension of Asterisk
C
What the dial plan should be?
Thanks
--
We never did too much talking anyway
So don't think twice, it's all right
-------------- next part --------------
An HTML attachment was
2009 Oct 26
1
state_interface backport issue
It's my understanding that the backport is available now in 1.4.
However, seem to be having some issues with it. Just wondering if I have
everything setup right.
I'm running 1.4.26.2 realtime.
queue_members:
`uniqueid` int(10) unsigned NOT NULL auto_increment,
`membername` varchar(40) default NULL,
`queue_name` varchar(128) default NULL,
`interface` varchar(128) default NULL,
2004 Nov 23
5
Fw: Gift for Mark Spencer
Why does this person have my e-mail address ?
----- Original Message -----
From: <markogift@astriholics.org>
To: <hackerwacker@cybermesa.com>
Sent: Tuesday, November 23, 2004 1:13 PM
Subject: Gift for Mark Spencer
> Hello everyone!
>
> We have been thinking about something that we could do for Mark
> Spencer as a holiday gift. We have decided to try to orgranize a