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

[j.williamson at connectfree.co.uk]

Oh... thanks...  :O) 

 

Sean McRobbie writes: 

> 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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>   
> 
> 
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