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Call Recording in Asterisk

If you want to do call recording in Asterisk, Mix(Monitor() is your friend that you can use in dialplan.

For example you want to record the calls coming on DID 1949 555 55555
exten => 19495555555,1,MixMonitor(${UNIQUEID}.ulaw)
same => n,Dial(SIP/101)

In another example if you want to record call on user extension 101
exten => 101,1,MixMonitor(${UNIQUEID}.ulaw)
same => n,Dial(SIP/101)

MixMonitor() records the audio from both directions of the phone call and writes it to a file on disk in one of the audio formats that Asterisk supports. You can see a list of the file formats that your version of Asterisk supports at the Asterisk CLI. The Extensions column identifies which file format extensions can be used in the recording filename:

*CLI> core show file formats
Format Name Extensions
—— —- ———-
gsm wav49 WAV|wav49
slin16 wav16 wav16
slin wav wav
adpcm vox vox
slin16 sln16 sln16
slin sln sln|raw
siren7 siren7 siren7
siren14 siren14 siren14
g722 g722 g722
ulaw au au
alaw alaw alaw|al|alw
ulaw pcm pcm|ulaw|ul|mu|ulw
ilbc iLBC ilbc
h264 h264 h264
h263 h263 h263
gsm gsm gsm
g729 g729 g729
g726 g726-16 g726-16
g726 g726-24 g726-24
g726 g726-32 g726-32
g726 g726-40 g726-40
g723 g723sf g723|g723sf
g719 g719 g719
23 file formats registered.

The syntax for MixMonitor() in the dialplan is as follows:
MixMonitor(filename.extension[,options[,command]])

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Asterisk one way audio issue

Are you having an audio issues in your Asterisk? 
Well it’s a common issue with PBX to have audio issues like one way audio or no audio. Sometimes only caller can hear remote party or remote party only can hear the caller. You must be wondering what causes this issue? This problem in audio is mainly because of the NAT issues. We recommend to use NAT with enabling 10000-20000 UDP ports on firewalls and also to enable natting on trunks

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How to limit the number of calls in asterisk

Learn how you can limit number of simultaneous calls on Asterisk based SIP Trunk. If you want to limit the number calls for your SIP peer or friend in Asterisk use call-limit in your trunk configuration.

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Asterisk time based routing

This is a very common requirement that route the calls to Voice-mail after office hours. Or you can transfer the calls to your cell phone after certain time say 6:00pm. In Asterisk you can also control the call location based on time and date.

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