On Thu, 2011-08-11 at 16:38 +0100, Paul Hayes wrote:
> > 2011/8/11 Ishfaq Malik mailto:i...@pack-net.co.uk>>
> >
> > On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote:
> > > Ah, now this is interesting as one of our clients had the same
> > problem the other day; in our case when
On 12/08/11 08:46, Ishfaq Malik wrote:
Have you seen it in any other versions of 1.8 or is it something that
has happened in the latest release?
I've not specifically seen this issue with other versions of Asterisk
but then I've never tried to replicate it. The only time I've seen this
with
On Fri, 2011-08-12 at 09:46 +0100, Paul Hayes wrote:
> On 12/08/11 08:46, Ishfaq Malik wrote:
> > Have you seen it in any other versions of 1.8 or is it something that
> > has happened in the latest release?
>
> I've not specifically seen this issue with other versions of Asterisk
> but then I've
Hello,
Is there a way to either store login/logout agent information in a database
or at least send an email when an agent logs in or out of a queue?
Thanks,
Michael
--
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-- Bandwidth and Colocation Provided by http://www.api-di
you can monitor queue_log file for ADDMEMBER or REMOVEMEMBER events. from that
point on, you can store them or take any other action.
the other way is to use AMI an monitor for Agent login/logoff events
On Aug 12, 2011, at 7:06 AM, Michael wrote:
> Hello,
>
> Is there a way to either store login
Is it possible to "butt in" on a call in progress and play a message to one
party, without disconnecting the call? (Anyone with fond [or not-so-fond]
memories of the old GPO payphones will remember the "pips" used to indicate
that a coin needed to be inserted to keep the call alive.)
Why do
On 08/11/2011 02:03 AM, Jim Boykin wrote:
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is th
12 aug 2011 kl. 14:51 skrev Kevin P. Fleming:
> On 08/11/2011 02:03 AM, Jim Boykin wrote:
>
>> We have difficulty setting up the incoming termination for our
>> clients. Both the ends are using asterisk. The problem is unless we
>> use fromuser at client end, it does not work properly as expect
Hi All,
Usually we need to queue calls coming from outside our system and for that
we use queues.conf, in this case we have a lot of employees that are using
POTS (Dahdi or zap channels) and we want to make them go by order since we
have limited outgoing lines, does anybody have a clue what to use
You can use "call-queueing" to accomplish this. When your employee dials
the number (555-1212 for demonstration purposes), instead of going directly
out, the call goes to /var/spool/asterisk/outgoing as an entry. When this
entry comes up, the employee gets a call-back/connect to his/her party. Y
Hi Danny,
Thnks for your response but I googled "call-queueing" with no success, are
your referring to the concept or a third party application or an Asterisk
function..., can you please specify?
On Fri, Aug 12, 2011 at 10:03 AM, Danny Nicholas wrote:
> You can use “call-queueing” to accomplish
Hi !
I have a python script that create and move .call files to
/var/spool/asterisk/outgoing
Sometimes...(in this case after 500 successfull calls) Asterisk don´t make
the calls and the .call files are in the "outgoing" forever...
Any Ideas?
I'm using Asterisk 1.4.22 (in 1.4.36 was the same behavi
On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:
>shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
Are both /var/tmp and /var/spool/asterisk/outgoing on the same
filesystem?
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_
-- Bandwidth
Yes, same server, same filesystem...
On Fri, Aug 12, 2011 at 12:26 PM, Roger Burton West wrote:
> On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:
>
> >shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
>
> Are both /var/tmp and /var/spool/asterisk/outgoing on the
On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote:
>Yes, same server, same filesystem...
I don't do Python, but a web search for shutil.move suggests that it
doesn't reliably use the "rename" syscall. Might be worth shelling out
to your system's mv command.
R
--
Also, keep in mind that the spooling mechanism has "mechanical limits" based
on processor speed, line capacity, etc. If I were doing 500 calls, I would
use sleep to space the starting of the calls (maybe 5 or 15 second
intervals).
-Original Message-
From: asterisk-users-boun...@lists.digi
Good Morning,
I have been researching this for a while, basically I'd like to have a
website with the following functionality:
• One-click call-in to show (after setting username, best-case
scenario: sign-in through Drupal, use that name for conference-call)
• Web-interface only (Flash/Flex, Javas
I made 500 calls but not simultaneously. My script checks that there
are no more
than 3 .call files in the "outgoing".
I change in my python script, now move file with os.system...
import os
os.system ("mv"+ " " + tmpFile + " " + callFile)
see what happens...
On Fri, Aug 12, 2011 at 12:40 PM, D
Another thought - when a call in /V/S/A/O fails, the file gets appended
with call info and retry occurs. You might want to write a second Python
script to check for and possibly purge failed call files.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.c
The .call file can connect an internal number to an outside number
Look at this sample
Channel: DAHDI/R1/5551212
MaxRetries: 2
# Retry in 5 min
RetryTime: 300
WaitTime: 45
Context: outgoing
Extension:100
Priority: 1
This sample call makes a call on DAHDI using Round Robin Group 1. If
On Fri, 12 Aug 2011, Danny Nicholas wrote:
Exten => 1234,2,AGI(makecall.agi,${EXTEN},${numtodial})
Makecall.agi
#!/bin/sh
echo "extension: $1" > call1.tmp
echo "maxtries: 3" >> call1.tmp
echo "retrytime: 300" >> call1.tmp
echo "Channel: DAHDI/R1/$2" >> call1.tmp
echo "Priority: 1" >> call1.tmp
Hey Danny thanks a bunch! I really appreciate that.
Thank you Steve!
On Fri, Aug 12, 2011 at 3:05 PM, Danny Nicholas wrote:
> The .call file can connect an internal number to an outside number
>
> Look at this sample
>
> Channel: DAHDI/R1/5551212
>
> MaxRetries: 2
>
> # Retry in
Hello,
Check if file is owned by "asterisk" user.
Also, don't directly create in to /var/spool/asterisk/outgoing/
Create in somewhere else first and then move file to outgoing folder.
Good luck.
On Fri, Aug 12, 2011 at 7:09 PM, Danny Nicholas wrote:
> Another thought – when a call in /V/S/A/O
In 1.8, somebody left a message that shows up like this
Remotely bridging SIP/Client.XX.XX.XX.125-00010456 and SIP/XX.XXX.XX.X-00010457
It could be also Local Bridging
The point is that this message should not print in the console unless
the verbose level reaches some level. Never at level zero. I
We are having a problem when trying to use originate or AMI to make a
call. We have an Asterisk 1.8.5.0 server which uses a SIP provider to
call the PSTN. When dialing from IP phones everything works fine. When
you try making the call with originate, AMI or a call file then the
remote pe
On 08/12/2011 03:23 PM, CDR wrote:
In 1.8, somebody left a message that shows up like this
Remotely bridging SIP/Client.XX.XX.XX.125-00010456 and SIP/XX.XXX.XX.X-00010457
It could be also Local Bridging
The point is that this message should not print in the console unless
the verbose level reach
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