On Friday 28 September 2012, Patrick Archibald wrote:
Hi,
Is there a way to move 100 .call files in to
/var/spool/asterisk/outgoing/ at once and have Asterisk call at
maximum 10 at a time?
Yes: Move them in batches of 10. Could be as simple as
last if ++$n_files 9;
if the script is in
Hello,
are there any known reasons why Asterisk would disconnect random calls ?
My server uses 1,5 GB out of 8 GB RAM
My server uses up to 35% CPU at peak
There are about 40 concurrent calls.
I have 300 RTP-ports available.
I just see the call ending, as if one of the connected parties hung
Le 28/09/2012 10:22, Jonas Kellens a écrit :
Hello,
Hi
are there any known reasons why Asterisk would disconnect random calls ?
My server uses 1,5 GB out of 8 GB RAM
My server uses up to 35% CPU at peak
There are about 40 concurrent calls.
I have 300 RTP-ports available.
I just see the
Hi,
According to http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip:
Quote:
If you place ipaddr before host (in the case of dynamic), you will never
load the public IP address of your sip device, as it will be overwritten when
host is encountered.
UnQuote.
From the latest Asterisk
On 28-09-12 10:31, Administrator TOOTAI wrote:
Le 28/09/2012 10:22, Jonas Kellens a écrit :
Hello,
Hi
are there any known reasons why Asterisk would disconnect random calls ?
My server uses 1,5 GB out of 8 GB RAM
My server uses up to 35% CPU at peak
There are about 40 concurrent calls.
I
On Fri, 2012-09-28 at 01:33 -0700, Vieri wrote:
Hi,
According to http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip:
[snip]
So it seems that the contrib directory and the asterisk.org wiki are
inconsistent and incomplete.
Of course I understand that these are 'contributed' files
Le 28/09/2012 10:40, Jonas Kellens a écrit :
[...]
are there any known reasons why Asterisk would disconnect random
calls ?
My server uses 1,5 GB out of 8 GB RAM
My server uses up to 35% CPU at peak
There are about 40 concurrent calls.
I have 300 RTP-ports available.
I just see the call
On 28-09-12 10:57, Administrator TOOTAI wrote:
Le 28/09/2012 10:40, Jonas Kellens a écrit :
[...]
are there any known reasons why Asterisk would disconnect random
calls ?
My server uses 1,5 GB out of 8 GB RAM
My server uses up to 35% CPU at peak
There are about 40 concurrent calls.
I have
--- On Fri, 9/28/12, Hans Witvliet aster...@a-domani.nl wrote:
how about the line:
`ipaddr` varchar(15) DEFAULT NULL,
Wonder how they try to squeeze an IPv6 address in it...
should be:
`ipaddr` varchar(50) DEFAULT NULL,
I think
`ipaddr` varchar(45) DEFAULT NULL,
should be
Hi list!
ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF
gets transmitted throughout the conference. I've tried Asterisk 10.7.1
from the official RPMs and 10.8.0 compiled from source.
I've confirmed that it's disabled via the CLI confbridge show profile
user
On 28/09/12 06:50 AM, Markus wrote:
Hi list!
ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF
gets transmitted throughout the conference. I've tried Asterisk 10.7.1
from the official RPMs and 10.8.0 compiled from source.
I've confirmed that it's disabled via the CLI
On 27/09/12 09:01 PM, Patrick Archibald wrote:
Is there a way to move 100 .call files in to
/var/spool/asterisk/outgoing/ at once and have Asterisk call at
maximum 10 at a time?
snip
I can certainly write a program to limit the number of simultaneous
outgoing calls but before I do that I
On 28/09/12 04:33 AM, Vieri wrote:
So it seems that the contrib directory and the asterisk.org wiki are
inconsistent and incomplete.
Of course I understand that these are 'contributed' files but they should be
proof-read by the Digium devs before packing them up into the official source
On 27/09/12 02:13 PM, Mehdi Rahimi wrote:
On Wed, Sep 26, 2012 at 11:31 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 26/09/12 05:35 AM, Mehdi Rahimi wrote:
I want to play music in my AGI while i am searching for a field in DB.
Actually during some processes in AGI i need to play
Hi Leif,
Am 28.09.2012 13:24, schrieb Leif Madsen:
Searching the issue tracker (hint, hint) does not return any
dtmf_passthrough issues other than this one[0], which doesn't look to be
related.
thanks for your reply.
Right, doesn't look related.
Is another channel connected to the
On 27/09/12 11:45 AM, Matt Hamilton wrote:
Date: Thu, 27 Sep 2012 10:23:35 +0200
From: lenz.lo...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR
I'd go for MyISAM and would set up a remote replica if data integrity is
important.
If
On 28/09/12 07:36 AM, Markus wrote:
Am 28.09.2012 13:24, schrieb Leif Madsen:
Is another channel connected to the conference receiving the DTMF? Is
that what you're intending? Because from my understand that is the
intention, and not simply to limit the DTMF from being in the conference
in the
On 28 Sep 2012, at 13:27, Leif Madsen leif.mad...@asteriskdocs.org wrote:
Generally the preferred method when you're doing this programatically anyways
is to use an external script through the Asterisk Manager Interface to
generate your calls.
Luckily, Russell Bryant has recently create
Hi everyone,
I am your friendly neighborhood developer here with a question that may
impact some of you.
Right now there is a small discussion occurring on the Asterisk
development mailing list about the expected behavior of music on hold
and AGI's stream file. Presently if you start music
On 28/09/12 08:11 AM, Aldo Bergamini wrote:
I am happy to hear that a new release of The Book is in the works!
That's good! I'd hate to be working on something no one wanted :)
I will have a look at Russell's script as soon as I am back at my work chair:
there is however something I am very
On 28/09/12 08:23 AM, Joshua Colp wrote:
I am your friendly neighborhood developer here with a question that may
impact some of you.
You're friendly? :)
Right now there is a small discussion occurring on the Asterisk
development mailing list about the expected behavior of music on hold
and
On 28 Sep 2012, at 14:24, Leif Madsen leif.mad...@asteriskdocs.org wrote:
That's good! I'd hate to be working on something no one wanted :)
;-)))
Oh heck ya. You can start up an Asterisk instance and just start doing things
with it via your programs. That's the immense power of AMI; it's
Leif Madsen wrote:
On 28/09/12 08:23 AM, Joshua Colp wrote:
I am your friendly neighborhood developer here with a question that may
impact some of you.
You're friendly? :)
3
Right now there is a small discussion occurring on the Asterisk
development mailing list about the expected
Tim Nelson wrote:
Is there a way to have Asterisk respond appropriately when receiving a DTMF
Flash event via SIP? I'm finding some WiFi SIP phones, specifically the
Quickphones QA-342 want to handle transfers/3-way calls by sending a DTMF Flash
event instead of handling it properly like
On 28/09/12 08:45 AM, Joshua Colp wrote:
Leif Madsen wrote:
I guess part of the question is; can you trigger it to be re-enabled
after the stream file?
Sure you can! You can use set music to start it going again as the next
command.
And that makes sense. I kind of knew the answer already,
Hi again Leif,
Am 28.09.2012 13:42, schrieb Leif Madsen:
OH! I just tested with a SIP softphone (X-Lite), and DTMF does not get
passed to the other users! In X-Lite I can hear the DTMF keypresses of
the users connected via PSTN (incoming via SIP), but when I hit a key in
X-Lite I can't hear
Markus wrote:
Snipped long results list
PSTN means that I've tested two times, from a regular landline and from
a mobile. Always calling to the providers DID which ends up in Asterisk
via SIP. In the case of ConfBridge there were always 2 participants in
the conference so that I could check if
Hello everybody,
i have a problem with asterisk 1.8 and Call Hold
My problem is that Asterisk don't send re-invite when i pick up the call from
hold.
I already insert canreinvite=no in all my sip channels, set dtmfmode=info in
sip.conf and my Dial() command don't insert option like t, T, h, H,
Hi Joshua,
Am 28.09.2012 15:56, schrieb Joshua Colp:
I think your results are sort of skewed. In the case of SIP - SIP if a
local bridge occurs things will optimize and you most likely won't see
DTMF related messages. They get passed through as packets and not fully
interpreted.
ah, ok! That
Markus wrote:
Hi Joshua,
Hola,
My suggestion is to take a step back further.
Just send incoming calls to the Read application and have it store the
received DTMF in a variable. Next step have it output what was received.
Ok, good idea, here are the results of Read() and SayDigits():
Thanks everyone for the guidance on my outgoing call question.
And I'm looking forward to Asterisk: The Definitive Guide 4e book too!
Thanks, PLA
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Am 28.09.2012 17:33, schrieb Joshua Colp:
Ok, good idea, here are the results of Read() and SayDigits():
snipped results to make this email manageable
How are you changing the DTMF for each provider? If you are merely
changing it using dtmfmode in sip.conf this may or may not change how
the
Markus wrote:
Am 28.09.2012 17:33, schrieb Joshua Colp:
Ok, good idea, here are the results of Read() and SayDigits():
snipped results to make this email manageable
How are you changing the DTMF for each provider? If you are merely
changing it using dtmfmode in sip.conf this may or may not
I want to put a call me now button on the web site that will place the
request into an asterisk call queue and then when an agent picks up the
call in the queue, place the outbound call to the customer.
The following AMI command works, but it calls the customer first, before
an agent is
On 27/09/12 02:13 PM, Mehdi Rahimi wrote:
I want to play music in my AGI while i am searching for a field in DB.
Actually during some processes in AGI i need to play music .
On Fri, 28 Sep 2012, Leif Madsen wrote:
It's been quite some time since I did this, so I can't give you a specific
On 9/28/2012 12:42 PM, Mitch Claborn wrote:
I want to put a call me now button on the web site that will place the
request into an asterisk call queue and then when an agent picks up the
call in the queue, place the outbound call to the customer.
The following AMI command works, but it calls
That approach only works if there are any agents that are not busy on a
call - I could pick one, take them out of the queue then connect the
call. If all agents are busy, I need to be able to insert the request
into the queue so that it gets processed in sequence with the inbound calls.
I was asked today if we could somehow have a trainee on the phone with a
supervisor conferenced in, but somehow have it so anything the
supervisor says is only heard by the trainee and not the customer.
Is there a feature like that?
--
On Fri, Sep 28, 2012 at 5:27 PM, Adam Moffett adamli...@plexicomm.net wrote:
I was asked today if we could somehow have a trainee on the phone with a
supervisor conferenced in, but somehow have it so anything the supervisor
says is only heard by the trainee and not the customer.
Is there a
Looking for ideas and comments on my strategy for getting a bit of
custom data into the CDR. It seems to work OK, but I'm open to better
and/or more robust ways to do it.
Problem: get the customerid of the caller from our application into the CDR
Approach:
Before the Queue() command, save
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