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

[Sean McRobbie]

I would recommend staying away from Trixbox. They are about as far away from Open Source as you can get.

For that, use FreePBX

Regards,
Sean McRobbie

----- Original Message -----
From: "j williamson" <j.williamson at connectfree.co.uk>
To: "Tayside Linux User Group" <dundee at lists.lug.org.uk>
Sent: Monday, 15 June, 2009 15:39:05 GMT +00:00 GMT Britain, Ireland, Portugal
Subject: Re: [dundee] Oh Asterisk and Sipgate why you mock me so....

Here's me thinking that finally I can help with something... but...  I used 
to use TRIXBOX, relatively easy but haven't used it since I moved home.  
Said to my wife, "but look what we can do with it", she said, "no!"  Maybe 
you might find the tutorial for TRIXBOX helpfull. 

James 


Sean McRobbie writes: 

> 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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