Asterisk/AGI Handle concurrent calls

Hi
If I have a SIP provider that gives me a SIP Account with a number that has many channels to handle multiple incoming calls, I want to know if Asterisk/AGI can handle those many concurrent calls (like above 10 calls)? Or do I need anything extra like voximal_interpreter?

Code in my dialplan below

[incoming]
exten => _x.,1,Answer
same=>2,Wait(2)
same=>3,Playback(/var/lib/asterisk/agi-bin/texttospeech/src/sound/welcome) //welcome message
same=>5,AGI(/var/lib/asterisk/agi-bin/callstream.php) //Record call via Google STT API
same=>6,Playback(/var/lib/asterisk/agi-bin/texttospeech/src/sound/outpusss11) //Playback output to caller

The effective code in your dialplan is actually only:

[incoming]
exten => _x.,1,Answer
same=>2,Wait(2)
same=>3,Playback(/var/lib/asterisk/agi-bin/texttospeech/src/sound/welcome) //welcome message

Asterisk doesn’t really care whether calls go to the same or different URIs. Any limit will be imposed externally.

Many many thanks for the reply.
Actually the numbering is 1,2,3,4,5 and not 1,2,3,5,6. Hope this is what you were expecting?

You may find it helpful to use 'same = n' instead of explicit priority numbering.

Also a 'Preformatted text' snippet of 'dialplan show x@y' can be helpful. What Asterisk thinks your dialplan says is more important than what you think it says :slight_smile:

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