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

[Daryll Strauss]

On Mon, 2004-08-30 at 09:07, Kevin Walsh wrote:
> That can be caused by the lack of QoS on your network.  You need to
> give priority to VoIP traffic - especially if you are saturating your
> network connectivity.

Ah, my router can do QoS, but I haven't turned it on. Another good thing
to play with. Thanks!

> You are looking for "canreinvite = no", which is a sip.conf setting.
> This will allow the endpoints to establish a direct link to one another,
> and remove Asterisk from the loop.  Of course, you can only remove
> Asterisk from the loop if Asterisk is no longer required.
> 
> 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.

That makes a lot of sense. In fact I don't have canreinvite, but I do
have t and T in my Dial, so that's why Asterisk is holding onto the
call. That makes perfect sense. Now I have to decide how important
transfers are or at least how well QoS really works.

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.

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)

Thanks for the details Kevin!!!

						- |Daryll