search for: tichi

Hi I am new on this forum. I am searching for a function in R which provides all partitions of a set, say for the set {1,2,3} you get {{1,2,3}} {1,{2,3}} {2,{1,3}} {3,{1,2}} {{1},{2},{3}} . The number of partitions of a set is given by Bellsche`s number. The number of possibilities of length (this is the number of subsets included in the partition) of the partition is given by the partition

Hi, in the Snom FAQ I found the following information: After staring up, the phone tries the URL given in the "Setting URL" of the phone. ... BTW this setting can also be set via DHCP. .... option tftp-server-name "http://192.168.0.9/snom200{mac}.htm" The documents used: FAQ-04-06-14-sf.pdf "Setting up DHCP for snom phones" FAQ-04-03-24-sf.pdf "How can I

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

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?

MixMonitor seems to work: -- User hit '*3' to record call. filename: auto-1250792853-24-22 == Begin MixMonitor Recording SIP/snom2-084c4ec8 /var/spool/asterisk/monitor/auto-1250792853-24-22.wav exists now. Recording a call without mixing fails. > User hit '*1' to record call. filename: wav,auto-1250793354-24-22,m TOUCH_MONITOR_OUTPUT is set to

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

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

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] --

Am Wed, May 24, 2023 at 09:40:18AM -0400 schrieb Steve Matzura: > > On 5/24/2023 7:49 AM, Stefan Tichy wrote: > > Am Tue, May 23, 2023 at 07:22:22PM -0400 schrieb Steve Matzura: > > > > > 1. Still can't register my phone > > > The username and password are correct. I don't know what else to try. > > You can start a sip trace from the asterisk

This was supposed to go to the list. I am now thoroughly confused. In the [voipms] stanza where endpoint is defined (type=endpoint), everything points to voipms. But in the [yealink] stanzas, I tried pointing everything to Steve, one item at a time, then both of them, and nothing changed. On 5/24/2023 10:00 AM, Stefan Tichy wrote: block quote Am Wed, May 24, 2023 at 09:40:18AM -0400 schrieb

Some unsuccessfull attempts to make console calls working. If a sip phone is called, the other side will hear nothing. If I try to record some sound the application will not finish. There is a sound file, but it is empty (0 bytes). "Record(${FILE}:gsm|10|30|skip)" is used in the dial plan. After hangup the following error messages show up: NOTICE[]: channel.c:1683 ast_set_read_format:

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

On 26-10-16 15:03, Dan Jenkins wrote: > > > On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > > I keep getting the following error when trying to connect to the > Asterisk server using AMI : > > $socket = fsockopen("tls://11.22.33.44 >

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:

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

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.

Hello, I wonder if anybody is using PJSIP realtime in production environment? I've started to play with it and encountered many problems. Here's my config: sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints extconfig.conf: [settings] ps_endpoints => pgsql,users,pjsip_endpoints_v pjsip_endpoints_v is postgresql view. 1. The biggest problem: if I have small number of endpoints

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 sure? The value for silencethreshold (140) is unusually large. -- Stefan Tichy ( asterisk3 at pi4tel dot de )

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 sure? Yes, im sure. I have looked at the time and

Hi Karsten, On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote: > .... The server sends Server Hello, Certificate, Server Key > Exchange and Server Hello Done. Something in that packet seems to be unacceptable for openssl 1.1.1d as it is compiled and configured for Buster. Certificate length, Digest algorithm, ... You my change the system default settings at the