Asterisk cmd Transfer

Source: Internet
Author: User
Tags unsupported
Transfer synopsis

Transfer caller to remote Extension


Transfer ([tech/] Dest [| options])

Requests the remote caller be transferred to a given extension.
If tech (SIP, iax2, local etc) is used, only an incoming call with the same channel technology will be transfered.
Note that for SIP, If you transfer before call is setup, a 302 redirect SIP message will be returned to the caller. it is possible to use 302 redirection to load-balance sip callamong several servers. A patch that fixes a few bugs is here: patch 9483.

The result of the application will be reported in the transferstatus channel variable:

Success transfer succeeded
Failure transfer failed
Unsupported transfer unsupported by Channel Driver

The option string contains contain the following character:

'J'-jump to N + 101 priority if the channel transfer attempt fails

Olle in Jan 2006: Transfer () is currently the only way to issue an outbound sip refer in the dial plan.
The transfer () application sends a refer, then totally ignores what is happening with the transfer. it does need a total rewrite for all VoIP channels and shoshould propably be considered a bad hack (read more ).

For older asterisk versionsThat still support priority jumping (1.0, 1.2 If enabled): If the transfer wasNotSupported or successful and there exists a priority n + 101, then that priority will be taken next.

Question: Where-should t for a sip refer-is this application of use, why not simply use dial (), maybe together with a local channel?

Answer: For example, a sip provider offering dids, you want callto go to a Colo box and then be transfered to other locations rather then waiting for the did provider to update routes, by using this you save headaches with 1 way audio due to tracking firewils not being able to track asyncronious sip and RTP paths, and call delay if you and the Colo are in different countries. downside to doing something like this is you incur a slightly increase call setup delay while the call is bounced between servers, but no one will ever notice since most people take a few rings to actually answer.

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