SIP provider questions.

[geoffrey mendelson]

On Jan 8, 2010, at 8:32 AM, Rami Addady wrote:

> Hi,
>
> 012.net  provide SIP trunk (minimum 4 lines) spikko.com provide SIP/ 
> IAX


Have you (or anyone else for that matter) gotten Spikko to work with  
asterisk? I signed up (it's free, why not :-) but can not get it to  
connect.
I get it to register, but calls never are connected to my asterisk  
system. I'm connected via 012 using an aDSL line and the normal BEZEQ  
Siemens
router.

If I use a softphone (with a Mac, so I can't use theirs so I use  
Zoiper), it registers, but the same thing happens. When I turn on  
STUN, it works and I can call it and it connects. Setting various  
versions of nat=yes, no nat at all, stun= (various servers) or no  
stun, asterisk registers but never connects.

my sip.conf:

register =>  <spikkousername>@d1.spikko.com

[d1.spikko.com]
type=friend
insecure=port,invite
host=d1.spikko.com
dtmfmode=rfc2833
canreinvite=no
secret=*************
username=spikkousername
context=spikko
port=5090
stunaddr=stun.zoiper.com:3478   ; tried with and without the port number
nat=yes

I know the bottom part is working, as if I change section name it  
fails to register with a bad password.

I have a context spikko in my dialplan.

Or does anyone know the IAX parameters? I can't find them with a web  
search or on the site.

Thanks, Geoff.


-- 
geoffrey mendelson N3OWJ/4X1GM
Jerusalem Israel geoffreymendelson at gmail.com
New word I coined 12/13/09, "Sub-Wikipedia" adj, describing knowledge  
or understanding, as in he has a sub-wikipedia understanding of the  
situation. i.e possessing less facts or information than can be found  
in the Wikipedia.