[dundee] Oh Asterisk and Sipgate why you mock me so....

[Sean McRobbie]

We have several Asterisk servers using sipgate too so if you need any further help let me know.

Regards,
Sean McRobbie

----- Original Message -----
From: "James Le Cuirot" <chewi at aura-online.co.uk>
To: dundee at lists.lug.org.uk
Sent: Monday, 15 June, 2009 11:50:14 GMT +00:00 GMT Britain, Ireland, Portugal
Subject: Re: [dundee] Oh Asterisk and Sipgate why you mock me so....

On Mon, 15 Jun 2009 11:28:50 +0100
Arron Finnon <afinnon at googlemail.com> wrote:

> I'm currently setting up an asterisk server with a sipgate number, i'm
> amazed i can write this email with blood in my eyes and the minor
> brain damage gained from banging my head of a brick wall.

Hi Arron,

You're in luck. I've been using sipgate for quite a while but I only
added Asterisk to the mix last week and it's been working fine! My
config is quite minimal so here it is. Note that I'm using the newer
syntax that doesn't require a "register" statement. I'm not sure which
version of Asterisk introduced this but I'm using 1.6.


[general]
context=default
srvlookup=yes
language=en
canreinvite=no
limitonpeers=yes

[sipgate]
type=friend
host=sipgate.co.uk
fromdomain=sipgate.co.uk
fromuser=8658235
callbackextension=8658235
defaultuser=8658235
secret=XXXXXXXX
insecure=invite

disallow=all
allow=g729
allow=gsm

[siemens]
type=friend
defaultip=192.168.1.201
permit=192.168.1.201
host=dynamic
secret=XXXXXXXX
qualify=yes
call-limit=2

disallow=all
allow=g729
allow=gsm
allow=alaw
allow=ulaw


Glancing at your config, I'm not sure what's missing. I know
insecure=invite is sufficient so you don't need insecure=very. My
situation may be a little simpler though because my server is also my
router so there's no NAT going on between Asterisk and sipgate. Still,
that shouldn't be an issue. I did have to open the RTP ports on the
firewall. I suspect you may have to forward them as well but you've
already done this. Maybe put your server in the DMZ to be sure. It's
not necessary to allow 5060 though unless you want people to be able to
call you directly without going via sipgate. You may want this since
sipgate blocks incoming SIP calls, except from select providers.

Hope this helps,
James

P.S. If you want, I can show you my really nifty trick for fading down
the volume when a call comes in and fading it up when the call ends. ;)
It takes multiple calls into account.

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