3 Feb
2012
3 Feb
'12
9:09 p.m.
We were in similar situation. Many years with Asterisk and then we were
forced to use ser - and we preferred Kamailio.
Now we do:
Kamailio has global IP address and clients register to it.
Kamailio forward all calls to Asterisk boxes using following:
ds_select_dst("1","4");#You can use many asterisk boxes this way
$sht(forw=>$ft)=$du; #this way I store used path
I used t_relay, instead of forward, because my Asterisks are with local IP.
Calls from Asterisk are send to Kamailio if they are to local user, or to
our SIP provider. There are no problems with calls from Asterisk to SIP
provider, even if Asterisk is behind NAT.
Asterisk accepts calls from SIP provider though registrar lines in
sip.conf. Asterisk can forward calls from our SIP provider to local users
in Kamailio.
I got problems with ACK and BYE. To solve them, I used
if(($td=="sip.name.of.kamailio.server.com")||($si=="IPofServer")){
$du=$sht(forw=>$ft);
}
On Fri, Feb 3, 2012 at 8:13 PM, Greg Mannie <greg(a)latigi.com> wrote:
Hello
After much reading I have come to the realization that after years of
using Asterisk I know very little about Sip.
I have my Kamailio box up, I have Asterisk 1.8.x running with realtime
working. I thought it would be just a case of registering SIP trunks from
my provider to the kamailio and registering our internal asterisk servers
to the kamailio.
Much of what I read talks about using Asterisk as the PSTN interface, but
that interface is through a sip trunk purchased from a provider. Won't
Kamailio be the PSTN gateway? The idea here is to pool all the sip trunks
from the various hosted asterisk solutions (VM running asterisk) and point
them all to a proxy to facilitate the aggregation of traffic.
I have been reading SIP tutorials and the mailing list archives. If
anyone has a sample config and perhaps a little direction it would be
highly appreciated.
Thank you
Greg
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