Displaying 12 results from an estimated 12 matches for "piafe".
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piaf
2005 Jan 27
1
computing roots of bessel function
I am not yet a R user but I will be soon.
I am looking for the R command and syntax to compute the roots of Bessel
function i.e. computing the z values that lead to Jnu(z)=0 where J is a
Bessel function or order nu.
May You help me ?
thanks in advance.
Dr Catherine COUTAND
Institut National de la Recherche Agronomique (INRA)
umr Physiologie Int?grative de l'Arbre Fruitier et Forestier
2019 Nov 08
0
[2.3.8] possible replication issue
...can help...
Regards.
Fabien.
+--------+
For technical details :
Container images are based on Arch Linux distro image.
Here is the configuration of Dovecot server :
dovecot -n
# 2.3.7.2 (3c910f64b): /etc/dovecot/dovecot.conf
# OS: Linux 4.19.66-coreos x86_64 ext4
# Hostname: mailstore-0.mailstore.piafe.svc.cluster.local
disable_plaintext_auth = no
doveadm_password = # hidden, use -P to show it
doveadm_port = 12345
login_greeting = PIAFE Mail Server Ready
mail_location = mdbox:/var/mail/%Ld/%Ln
mail_plugins = " notify replication quota"
namespace inbox {
hidden = no
inbox = yes
lis...
2015 Apr 07
0
Asterisk 13 very high trasnlation time between codecs
Hi all.
I've installaed a PIAF with Asterisk 13.0 on a ESX virtual machine on a DELL
node.
As usual I loaded g729 codec in modules folder but I notice very high
translation time.
Over several PIAF with Asterisk 1.8.X (always in a virtual environment) I
always had for example 601 between G729 and ALAW or ULAW while in this case
I have 15000.
Even recalculating with different values (eg:
2011 Oct 20
1
10.0 CallerID question
Hi List,
Another dumb conversion question (I hope). I installed 10.0
and copied my 1.4 configuration files over. With a few tweaks everything
works great except for 1 feature that I specifically went to 10.0 for. When
I do an attended transfer, I still get the receptionists caller ID on the
transferred phone instead of the incoming callerID. My assumption is that
there is some
2015 Jun 15
1
small homebrew pbx
James Cass wrote:
> I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php for $30, and it works great for a home install. Very low power draw as well.
>
> James Cass <http://goog_987864563>
> jcass78 at gmail.com <mailto:jcass78 at gmail.com>
The JS-200 runs a very old ( 1.4 ) version of Asterisk
I have set up more than 30 nodes using the HP thin
2009 Nov 05
3
programming phones
I have question thats not really about astrisk but I figure you guys are doing this sort of thing.
We use Aastra 6757i phones. there is some support for XML. the question is how would i go about learning to customize these phones?
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2008 Oct 28
1
Multiline Analog Setup
What is involved in provisioning Asterisk to use a multiline analog service from our local telco?
I will only have one twisted pair entering in on a OpenVox card but am not sure how Asterisk
interprets and deals with two incoming calls and/or two outgoing calls?
Thanks!
jlc
2009 Apr 23
2
Asterisk on Mac OS X
Hello list.
I posted this over on the Biz section but some of the members thought
I might find more people running Asterisk on the Mac over here.
Here's my question:
I have looked at PHLink and PhoneValet and neither seem to be able to
do what I need, so I am looking at Asterisk.
What I want to do is allow callers to call a our phone line and
unsubscribe their phone number from our call
2011 Jun 21
4
call paging interrupts call when using Mitel 5224
Is anybody using Mitel phones? It appears that when you page a Mitel phone
using asterisk's MeetMe, the paged phone will hang up the call its on to
take the page. Thanks in advance.
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2015 Jun 15
2
small homebrew pbx
> I don't know this 'translates' to Italy, but this is what I would advise
> somebody in the US to consider, assuming you have a reliable Internet
> connection.
>
> 0) I hope you mean you want to run Asterisk at home instead of 'Asterisk
> at Home.' A at H was an ancient distribution from around 2005.
>
> 1) Rent a DID (a 'PSTN number')
2010 Jun 06
1
Error of FreePBX after installing from Yum Repository of Asterisk
Hi Guys,
Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When
trying to dial a number, I get this:
tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/
op_server.pl line 3367.
Use of uninitialized value in concatenation (.) or string at
/var/www/html/panel/op_server.pl line 3372.
Use of uninitialized value in pattern match (m//) at
2009 Jul 23
1
[PATCH server] changes required for fedora rawhide inclusion.
Signed-off-by: Scott Seago <sseago at redhat.com>
---
AUTHORS | 17 ++++++
README | 10 +++
conf/ovirt-agent | 12 ++++
conf/ovirt-db-omatic | 12 ++++
conf/ovirt-host-browser | 12 ++++