Elastix-2.4.0-Stable-x86_64-bin-04feb2013.iso
CLI:
asterisk -r
asterisk -rvvvvvv
watch -n1 asterisk -rx \"sip show peers\"
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DIALPLANE:
/etc/asterisk/extentions.conf
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TFTP:
SolarWinds TFTP Server
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http://downloads.snom.com/fw/snomPA1-8.7.3.25-SIP-f.bin
http://provisioning.snom.com/download/fw/snomPA1-8.7.3.25-SIP-f.bin
tftp://192.168.1.10/snomPA1.bin
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http://wiki.snom.com/Snom360/Firmware/V8
snom360 8.7.3.25 --> Download --> Rename to snom360.bin --> Copy into TFTP Server Download Folder
http://downloads.snom.com/fw/snom360-8.7.3.25-SIP-f.bin
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VAD---Voice Activity Detection
CNG---Comfort Noise Generator
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/etc/asterisk/chan_dahdi.conf
busydetect=yes
busycount=6
service asterisk restart
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BUSY LINE:
^^^^^^^^^^^^^^^
busydetect=yes
busycount=0
service asterisk restart
X-Lite 3/Eyebeam > Advanced > Send SIP keep-alives & Use rport
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rxgain=3.0
txgain=3.0
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CODEC:
^^^^^^^^^^^^^^^
g711
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ECHO:
^^^^^^^^^^^^^^^
echocancel = yes
echocancelwhenbriged = no
echotraining = yes
service asterisk restart
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/etc/asterik/logger.conf
var/log/asterisk/full
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FARSI SOUNDS:
^^^^^^^^^^^^^^^
/etc/asterik
chan_dahdi.conf
queues_custom.conf
sip_custom.conf
sip_general_custom.conf
language=pr
var\lib\asterisk\sounds\pr
say.conf
etc/asterisk
mode=new
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LANGUAGE CODE:
^^^^^^^^^^^^^^^
any extention!
pr
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DEFAULT SOUND:
^^^^^^^^^^^^^^^
/var/spool/asterisk/voicemail/default
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password:
^^^^^^^^^^^^^^^
vi /etc/amportal.conf
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/var/www/backup
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custom context
http://svn.freepbx.org/modules/release/2.9/
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ring group
ring strategy
Extension Quick Pick
Failover Identity
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; Auto-generated by /usr/sbin/hardware_detector
[trunkgroups]
[channels]
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=3.6
txgain=3.6
callgroup=1
pickupgroup=1
relaxdtmf=yes
;Uncomment these lines if you have problems with the disconection of your analog lines
busydetect=yes
busycount=6
immediate=no
#include dahdi-channels.conf
#include chan_dahdi_additional.conf
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FLASH OPERATOR PANEL CUSTOMIZATION:
^^^^^^^^^^^^^^^
var/lib/asterisk/bin/retrieve_op_conf_from_mysql.pl
var/www/html/panel/op_style.cfg
var/www/html/panel/op_buttons_additional.cfg
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sip reload #Reload sip.conf (added after 0.7.1 on 2004-01-23)
dialplan reload #
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HOW TO CHECK CODEC INSTALLED?:
^^^^^^^^^^^^^^^
asterisk -rx "core show codecs"
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SHOW ACTIVE SIP CHANNELS:
sip show channels
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RESOLVING AUDIO PROBLEMS:
^^^^^^^^^^^^^^^
1 - Codec: ulaw, alaw
2 - /var/log/asterisk
3 - ping
4 - rxgain
5 - CPU / Memory Usage?
6 - Virtualization!
7 - delay
8 - Eyebeam > Options > AGC
9 - HDD
10 - SNOM > Settings > codec priority list
11 - Wireshark - Data Loss
12 - FreePBX > Tools > Asterisk SIP Settings > unselect ALL codecs > adding codecs ONE AT A TIME in order of preference
13 - sound card
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SNOM PA1: LED INDICATIONS WHEN APPLICATION IS RUNNING:
^^^^^^^^^^^^^^^
Red LED Green LED
Connect the device On Off
Web interface available On On
Ready Off On
Busy Off Blinking
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SNOM PA1:
^^^^^^^^^^^^^^^
POWER ADAPTER : 5V 2A 10W
SNOM PA1: MIC in & LINE out
***SNOM PA1 LINE OUT IS BALANCED, SINGLE MONO***
***3.5 mm connector***
- 3.5mm mono :Ground mono
- 3.5mm stereo :Ground Right Left
---
SPK impedance = 150 Ohm
---
MIC power supply = 3.3V
RL (Input R amplifier) = 2 KOhm
Sensitivity = -42 dB (+-2dB)
---
***Microphone input must be differential and without grounding***
***TWISTED-PAIR***
G.711 codec with 20ms packet length used
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SNOM : NOISE !!!
^^^^^^^^^^^^^^^
Settings > PA1 AMP Gain = 0
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SNOM 360: RESET
^^^^^^^^^^^^^^^
1 - settings button
2 - scroll down to reset values
3 - Press the check button
