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Action: Login
Username: hello
Secret: world
Response: Success
Message: Authentication accepted
Now you’re ready to originate your call. We’re doing essentially the same thing we did
with the call file, only this time using the AMI:
Action: Originate
Channel: PJSIP/SOFTPHONE_A
Context: sets
Exten: 103
Priority: 1
You should hear SOFTPHONE_A ringing. As soon as you answer it, a call will be placed
to SOFTPHONE_B.
AMI is no longer involved in what’s going on. You can disconnect and the call will
continue (leave the call up for now, as we’re going to work with the in-progress call
next).
Action: Logoff
Response: Goodbye
Message: Thanks for all the fish.
Connection closed by foreign host.
Example Usage
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If you’ve already hung up the call, that’s no problem. You’ll just need to re-establish
the call, which of course you can do simply by calling one extension from the other
(101 to 103, or whatever you want).
Redirecting a Call
Redirecting (or transferring) a call from the AMI is another feature worth mention‐
ing. The Redirect AMI action can be used to send one or two channels to any other
extension in the Asterisk dialplan. If you need to redirect two channels that are
bridged together, do them both at the same time. Otherwise, once one channel has
been redirected, the other will be hung up.
An important thing to understand about Asterisk channels is that they don’t exist
until a call is in progress. The name we all think of as the channel name (e.g., SOFT
PHONE_A) is not in fact the channel name, but merely a reference to data that is used to
create a channel. The naming of a channel takes place when a call is originated (which
is when the channel is actually created). What all this means is that you have to deter‐
mine the full name of the channel before you can act on it.
Originate a call, and then review the Event: Newchannel and you will see the channel
name under the Channel: header.
Action: Originate
Channel: PJSIP/SOFTPHONE_A
Context: sets
Exten: 103
Priority: 1
Response: Success
Message: Originate successfully queued
Event: Newchannel
Privilege: call,all
Channel: PJSIP/SOFTPHONE_A-00000013
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum: <unknown>
CallerIDName: <unknown>
ConnectedLineNum: <unknown>
ConnectedLineName: <unknown>
Language: en
AccountCode:
Context: sets
Exten: s
Priority: 1
Uniqueid: 1538939479.29
Linkedid: 1538939479.29
The Newchannel event will provide the name of the created channel, which in this
example is PJSIP/SOFTPHONE_A-00000013.
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Chapter 17: Asterisk Manager Interface and Call Files
You will need to keep track of these channel names if you wish to properly perform
actions on calls in progress. Once the call ends, the channel is destroyed. A new call
using the same endpoint will be assigned a different channel name. One channel defi‐
nition can support multiple calls (for example, multiple calls to a phone are possible),
and this is why the channel name is different from the channel definition.
You can redirect a single channel (the other channel will be disconnected):