[Asterisk-Users] How does call routing actually work with SIP?

[Kevin Walsh]

Daryll Strauss [daryll at daryll.net] wrote:
> On Mon, 2004-08-30 at 09:07, Kevin Walsh wrote:
> > Asterisk will remain in the loop if you have specified "t" or "T" in
> > your Dial() command, as it will need to listen for the hash key.  It
> > will also remain in the loop if you're recording the audio stream
> > using Monitor(), or whatever.
> >
> Now if my phone were really smart it would let me reinvite back to
> Asterisk somehow when I asked to do the transfer, but that would require
> smarts in the phone/Sipura case which I don't know if that exists.
>
You don't need "T" or "t" to transfer most of the time.  If you have
a SIP phone then it'll probably have a transfer facility anyway.  If
you're using an analogue phone on an ATA or a Sipura FXS then just
use the flash key.

> 
> By the way, people have been asking about echo. Other than this one test
> with the busy network I've heard no echos on my Sipura.
> 
> One thing the Sipura can't do, that I'd like is identify incoming
> distinctive ring. I have two numbers on the PSTN which differentiate by
> distinctive ring and I'd like Asterisk to handle them differently. I
> asked Sipura support and they said they can't do it yet. (Maybe that
> means a later firmware) 
> 
Yes - apparently that's due shortly, in a firmware upgrade.

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