Hi,
Spikko work also using IAX. You can find more information in this blog:
http://gilpalmon.com/2008/08/25/spikko-asterisk-free-iax2-to-pstn-in-israel/
Rami
geoffrey mendelson wrote:
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 geoffreymendel...@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.
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