VoIP: Difference between revisions
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[[Category:Lab]] |
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__TOC__ |
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<br /> |
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= Asterisk = |
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Installing Asterisk in Ubuntu |
Installing Asterisk in Ubuntu |
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sudo apt-get install asterisk |
sudo apt-get install asterisk |
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https://wiki.asterisk.org/wiki/display/AST/Installing+AsteriskNOW |
https://wiki.asterisk.org/wiki/display/AST/Installing+AsteriskNOW |
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= LAN Installation = |
== LAN Installation == |
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This section will allow you to : |
This section will allow you to : |
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Host calls locally |
Host calls locally |
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= |
== Conference Call == |
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Source [https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration wiki.asterisk.org] |
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Backup original file: |
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sudo mv /etc/asterisk/confbridge.conf /etc/asterisk/confbridge.conf.orig |
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Create & Edit the new file: |
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sudo nano /etc/asterisk/confbridge.conf |
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<pre> |
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[general] |
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; comments are preceded by a comma |
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; |
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; the general section is blank |
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; |
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[default_bridge] |
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type=bridge |
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; Bridge Profile options go here |
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myoption=value |
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myoption2=othervalue |
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; |
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[default_user] |
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type=user |
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; User Profile options go here |
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myoption=value |
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myoption2=othervalue |
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; |
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[sample_menu] |
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type=menu |
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; Conferece Menu options go here |
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DTMF=function |
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otherDTMF=otherFunction |
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; |
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[fancybridge] |
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type=bridge |
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max_members=20 |
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mixing_interval=10 |
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internal_sample_rate=auto |
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record_conference=yes |
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[fancyuser] |
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type=user |
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music_on_hold_when_empty=yes |
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music_on_hold_class=default |
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announce_user_count_all=yes |
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announce_join_leave=yes |
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dsp_drop_silence=yes |
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denoise=yes |
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pin=456 |
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[fancymenu] |
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type=menu |
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*=playback_and_continue(conf-togglemute&press&digits/1&silence/1&conf-leave&press&digits/2&silence/1&add-a-caller&press&digits/3&silence/1&conf-decrease-talking&press&digits/4&silence/1&reset-talking&press&digits/5&silence/1&increase-talking&press&digits/6&silence/1&conf-decrease-listening&press&digits/7&silence/1&conf-reset-listening&press&digits/8&silence/1&conf-increase-listening&press&digits/9&silence/1&conf-exit-menu&press&digits/0) |
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*1=toggle_mute |
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1=toggle_mute |
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*2=leave_conference |
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2=leave_conference |
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*3=dialplan_exec(addcallers,1,1) |
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3=dialplan_exec(addcallers,1,1) |
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*4=decrease_listening_volume |
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4=decrease_listening_volume |
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*5=reset_listening_volume |
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5=reset_listening_volume |
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*6=increase_listening_volume |
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6=increase_listening_volume |
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*7=decrease_talking_volume |
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7=decrease_talking_volume |
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*8=reset_talking_volume |
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8=reset_talking_volume |
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*9=increase_talking_volume |
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9=increase_talking_volume |
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*0=no_op |
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0=no_op |
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[addcaller] |
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exten => 1,1,Originate(SIP/otherpeer,exten,conferences,100,1) |
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[conferences] |
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exten => 100,1,ConfBridge(1234) |
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</pre> |
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Restart asterisk to load new files. |
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== PSTN Calling == |
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Source: [https://www.howtoforge.com/how-to-install-asterisk-for-your-first-pbx-solution howtoforge.com] |
Source: [https://www.howtoforge.com/how-to-install-asterisk-for-your-first-pbx-solution howtoforge.com] |
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{{UC}} |
{{UC}} |
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== Addons == |
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= Troubleshooting = |
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{{UC}} |
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== Troubleshooting == |
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Asterisk Console: |
Asterisk Console: |
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sudo asterisk -rvvv |
sudo asterisk -rvvv |
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sudo asterisk -rvvvvvv |
sudo asterisk -rvvvvvv |
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<br /> |
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;References |
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<references/> |
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<br /> |
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<br /> |
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<br /> |
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{{DISQUS}} |
Latest revision as of 01:11, 6 August 2017
Asterisk
Installing Asterisk in Ubuntu
sudo apt-get install asterisk
Standalone installation:
Download the latest ISO from below link & create a VM from it:
http://www.asterisk.org/downloads/asterisknow https://wiki.asterisk.org/wiki/display/AST/Installing+AsteriskNOW
LAN Installation
This section will allow you to :
Host calls locally Access mail boxes
We will be editing the following files:
sip.conf extensions.conf voicemail.conf
These files are located in:
/etc/asterisk
Making a backup of the 3 files:
sudo mv /etc/asterisk/sip.conf /etc/asterisk/sip.conf.orig sudo mv /etc/asterisk/extensions.conf /etc/asterisk/extensions.conf.orig sudo mv /etc/asterisk/voicemail.conf /etc/asterisk/voicemail.conf.orig
- sip.conf
sudo nano /etc/asterisk/sip.conf
[general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=192.168.1.0/255.255.255.0 [7001] type=friend host=dynamic secret=123 context=internal [7002] type=friend host=dynamic secret=456 context=internal
- extensions.conf
sudo nano /etc/asterisk/extensions.conf
[internal] exten => 7001,1,Answer() exten => 7001,2,Dial(SIP/7001,60) exten => 7001,3,Playback(vm-nobodyavail) exten => 7001,4,VoiceMail(7001@main) exten => 7001,5,Hangup() exten => 7002,1,Answer() exten => 7002,2,Dial(SIP/7002,60) exten => 7002,3,Playback(vm-nobodyavail) exten => 7002,4,VoiceMail(7002@main) exten => 7002,5,Hangup() exten => 8001,1,VoicemailMain(7001@main) exten => 8001,2,Hangup() exten => 8002,1,VoicemailMain(7002@main) exten => 8002,2,Hangup()
- voicemail.conf
sudo nano /etc/asterisk/voicemail.conf
[main] 7001 => 123 7002 => 456
Reloading Asterisk to apply the configuration
sudo asterisk -rx reload
OR
sudo /etc/init.d/asterisk restart
Conference Call
Source wiki.asterisk.org
Backup original file:
sudo mv /etc/asterisk/confbridge.conf /etc/asterisk/confbridge.conf.orig
Create & Edit the new file:
sudo nano /etc/asterisk/confbridge.conf
[general] ; comments are preceded by a comma ; ; the general section is blank ; [default_bridge] type=bridge ; Bridge Profile options go here myoption=value myoption2=othervalue ; [default_user] type=user ; User Profile options go here myoption=value myoption2=othervalue ; [sample_menu] type=menu ; Conferece Menu options go here DTMF=function otherDTMF=otherFunction ; [fancybridge] type=bridge max_members=20 mixing_interval=10 internal_sample_rate=auto record_conference=yes [fancyuser] type=user music_on_hold_when_empty=yes music_on_hold_class=default announce_user_count_all=yes announce_join_leave=yes dsp_drop_silence=yes denoise=yes pin=456 [fancymenu] type=menu *=playback_and_continue(conf-togglemute&press&digits/1&silence/1&conf-leave&press&digits/2&silence/1&add-a-caller&press&digits/3&silence/1&conf-decrease-talking&press&digits/4&silence/1&reset-talking&press&digits/5&silence/1&increase-talking&press&digits/6&silence/1&conf-decrease-listening&press&digits/7&silence/1&conf-reset-listening&press&digits/8&silence/1&conf-increase-listening&press&digits/9&silence/1&conf-exit-menu&press&digits/0) *1=toggle_mute 1=toggle_mute *2=leave_conference 2=leave_conference *3=dialplan_exec(addcallers,1,1) 3=dialplan_exec(addcallers,1,1) *4=decrease_listening_volume 4=decrease_listening_volume *5=reset_listening_volume 5=reset_listening_volume *6=increase_listening_volume 6=increase_listening_volume *7=decrease_talking_volume 7=decrease_talking_volume *8=reset_talking_volume 8=reset_talking_volume *9=increase_talking_volume 9=increase_talking_volume *0=no_op 0=no_op [addcaller] exten => 1,1,Originate(SIP/otherpeer,exten,conferences,100,1) [conferences] exten => 100,1,ConfBridge(1234)
Restart asterisk to load new files.
PSTN Calling
Source: howtoforge.com
This section is under construction. |
Addons
This section is under construction. |
Troubleshooting
Asterisk Console:
sudo asterisk -r
Debug Info:
sudo asterisk -rvvv sudo asterisk -rvvvvvv
- References
{{#widget:DISQUS
|id=networkm
|uniqid=VoIP
|url=https://aman.awiki.org/wiki/VoIP
}}