Displaying 20 results from an estimated 55 matches for "pi4tel".
Did you mean:
intel
2005 May 05
5
snom mass deployment (probably off topic)
Hello
Although not stictly a asterisk issue, any help would be apreciated.
Firstly a few notes on the snom 360, which I have had on a test bed
for the last week. Its a great phone, with a good user interface,
both physically and its web based one.
At its lastest firmware it does have a few quirks, with regards to the
way it handles usernames and passwords on the physical interface.
These have
2015 Jan 26
2
asterisk 11.14 - voicemail incorrect duration
Hi all,
i use asterisk 11.14.0 and I suspect that the voicemail application
counts the time wrong.
In my voicemail.conf:
[general]
minsecs=3
maxsilence=5
format=wav
maxsecs=180
silencethreshold=140
[...cut..]
In the asterisk-cli:
[Jan 26 15:23:49] -- Executing [s at macro-voicemail:77]VoiceMail("SIP/XY-0005175a", "aNumber,su") in new stack
[Jan 26 15:24:04] --
2020 Apr 30
2
SIP TLS not working, Asterisk 16.9.0
Hi,
I have problems with SIP via TLS. Asterisk works as a client. The TCP
connection is established, followed by a client hello from Asterisk to
the server. The server sends Server Hello, Certificate, Server Key
Exchange and Server Hello Done.
Than Asterisk sends back a Alert (Level: Fatal, Description Handshake
Failure). The following line appears in the log:
ast_iostream_start_tls: Problem
2006 Mar 22
2
Asterisk perms in manager.conf
Hi,
can someone sched a light what exactly mean the read write permissions
in manager.conf?
[public]
secret = private
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/255.255.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Lets say I want some users to use dial through manager interface. But
don't want to allow them to run asterisk commands?
2009 Aug 20
1
Asterisk 1.6.2.0-beta4 - Monitor / MixMonitor Recording
...ny new file.
Dial options are "twhx" and in features.conf there is:
[featuremap]
automon => *1
automixmon => *3
Is there anything else to follow to get monitoring working for
asterisk 1.6 or is it a bug in the 1.6.2.0-beta4 ?
Thanks in advance
--
Stefan Tichy ( asterisk2 at pi4tel dot de )
2015 Sep 16
4
Realtime Voicemail MWI
Greetings All, Regarding this archived post.
http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html
Did anyone ever find an solution to this? I've got a new box running
13.3.0 with the exact same issue.
For those that don't read the link.
I've got SIP Peers in realtime. All with a mailbox set. 98% of the time,
These are loaded into asterisk without
2004 Sep 27
3
chan_capi, Eicon Diva server BRI, kernel 2.6?
Hi list,
Does chan_capi work with kernel 2.6? The Eicon Diva server card loads
fine judging from /var/log/messages but Asterisk gives an error when
trying to load the chan_capi module. I'm using chan_capi-0.3.5,
zaptel-1.0.0, libpri-1.0.0 and asterisk-1.0.0 on a Fedora box with
kernel 2.6.8-1.584. Zaptel and ilbpri work fine as does *. I have seen a
msg that may be related and don't know
2005 Mar 17
3
Compilation problem chan_capi and Eicon Diva 4Bri
Hi *,
I want to integrate the Eicon Diva 4Bri Card to Asterisk.
Eicon drivers and capi is installed. I use the latest dev version from
eicon compiled and installed for my fedora 2 system.
I found the chan_capi for asterisk from www.junghanns.net. Also loaded
the patch and applied to the chan_capi source tree.
I changed the Makefile to include the capi20.h from eicon:
2004 Nov 21
4
Snom 190 - dhcp - settings_server
..."Setting up DHCP for snom phones"
FAQ-04-03-24-sf.pdf "How can I update a snom phone?"
The phone used is a snom 190 (snom190-SIP 3.52e).
If I use the webinterface to insert the URL it works fine, but I am
not able to set this URL using dhcp.
--
Stefan Tichy <asterisk@pi4tel.de>
2006 Jun 15
7
Executing a Function from AGI
Hmmm. Not having much luck with this. I'm trying to call the DUNDILOOKUP function and assign it to a variable in an AGI script.
I've tried setting with EXEC CMD and with SET VARIABLE. In both cases, it's treating DUNDILOOKUP literally, rather than calling a funciton.
I've tried this:
EXEC "Set" "DIALPATH=${DUNDILOOKUP(2944093|180net)}"
and also:
SET VARIABLE
2008 Nov 02
5
Ztdummy and Asterisk
Hi,
I have installed Asterisk 1.4.20 on Debian Etch. The server has no telephony card
installed, but I have anyhow installed Zaptel (Zaptel-1.4.9) in order to be able to use MeetMe.
The Zaptel modules load normally. I obtain the following prompts:
kerplunk:/# /etc/init.d/zaptel start
Loading zaptel framework: done.
Waiting for zap to come online...OK
Loading zaptel hardware modules: ztdummy.
2015 Jan 26
0
asterisk 11.14 - voicemail incorrect duration
..., Jan 26, 2015 at 04:37:23PM +0100, Dominique Haeber wrote:
> So, from 15:24:04 to 15:24:10 there are 6 seconds. But asterisk only
> count 2. What can be the reason? It is not silence.
Are you sure? The value for silencethreshold (140) is unusually large.
--
Stefan Tichy ( asterisk3 at pi4tel dot de )
2015 Jan 27
1
asterisk 11.14 - voicemail incorrect duration
Hi Stefan,
Stefan Tichy <asterisk3 at pi4tel.de> schrieb am Mon, 26. Jan 23:56:
> Hi Dominique
>
> On Mon, Jan 26, 2015 at 04:37:23PM +0100, Dominique Haeber wrote:
>
> > So, from 15:24:04 to 15:24:10 there are 6 seconds. But asterisk only
> > count 2. What can be the reason? It is not silence.
>
> Are you...
2020 May 01
0
SIP TLS not working, Asterisk 16.9.0
...is compiled and configured for Buster.
Certificate length, Digest algorithm, ...
You my change the system default settings at the bottom of
"/etc/ssl/openssl.cnf", restart asterisk and try again. Keep in
mind that this will affect the whole server.
--
Stefan Tichy ( asterisk3 at pi4tel dot de )
2023 May 24
0
Problems Solved, two left
...ated. You might have to change
the phone configuration.
> [yealink]
> type = aor
> contact = sip:Steve at 192.168.1.185
There should be no "contact" parameter for a phone. The phone sends
the required information with the register request.
--
Stefan Tichy ( asterisk3 at pi4tel dot de )
2004 Apr 02
3
cron job to reboot GS101
Does any one regularly reboot GS101? It sometimes lost registration with
* and needs to be reboot.
What is the best way to do it by cron?
David Kwok
2004 May 21
4
dial application - continue in context
Hi All,
I'm tring to do some DB operations before and after a call. I see the
'g' option in dial to continue in context if the destination hangs up,
but what if the originator hangs up?
Basically I do a DB get/put before the call is placed. After the call is
completed I want to do another get/put; however the dial application
dies when the originator hangs up.
Any way to get around
2004 May 25
1
Unable to create channel of type 'CAPI'
Since upgrading from stable to latest cvs I can't place CAPI calls (AVM
Fritz/chan_capi-0.3.1)
Did I miss something that has to be changed in configfiles?
Also tried to recompile chan_capi which run into an error.
capi info shows me:
Contr1: 2 B channels total, 2 B channels free.
Any suggestions to these logfile snippets
jo
----------------------------
* to ISDN
-- Accepting
2004 May 31
2
Users in MySQL
I've just compilied th latest CVS of * with USE_MYSQL_FRIENDS enabled ("1"). During
startup * tells me that it connects to the db, so this should be fine.
Nevertheless I don't see any users from the db when I run "sip show users" or "iax2 show
users" although I configured some.
It is also not possible to call them.
Any hints?
2004 Jun 28
1
Asterisk & Festival, not a happy couple
Hello,
I'm in the process of trying to get Festival to work with Asterisk. I
followed the install process at
http://www.voip-info.org/wiki-Asterisk+festival+installation. To get the
Festival to compile I had to add the patch described in the comments.
Once added, Festival and the Speech tools compiled without error.
How ever, when ever I try to call the test extension, I get a busy