[asterisk-app-dev] Accessing dialplan functions and SIP headers via ARI

[Alistair Cunningham]

On 18/12/14 22:50, Phil Mickelson wrote:
> Are you asking about the actual caller id or an internal value?  If
> caller id (the caller's telephone number) can't you get that from the
> StasisStart event?  I get it when the channel is created by accessing a
> variable I call "stasis" like this:  stasis.channel.caller.number.  If
> the caller allows their number to be visible I can see it just fine.
>
> Otherwise, if you're looking for a unique number why not just use the
> channel.id <http://channel.id> value?
>
> Sorry if I don't understand the question correctly.
>
> BTW, I'm using Asterisk 13 and the Node ARI interface.
>
> Phil M

The SIP headers we need to access are:

Alert-Info
Call-ID
Call-Info
Contact
Diversion
P-Asserted-Identity
Proxy-Authorization
Remote-Party-ID
X-Enswitch-Conference-ID
X-Enswitch-Conference-Music
X-Enswitch-Conference-Type
X-Enswitch-External
X-Enswitch-Prepend-Caller
X-Enswitch-RURI
X-Enswitch-Source
X-Enswitch-Spy-Channel
X-Enswitch-Spy-Options
X-Enswitch-Uniqueid

The X-Enswitch-* ones are used to pass supplemental information when 
forwarding calls between Asterisk machines in a cluster.

-- 
Alistair Cunningham
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