When your Asterisk Box sends calls back to Kamailio do IP auth on the
Kamailio and let traffic pass because its from a trusted source.
On 6/17/13 4:27 PM, Mickael MONSIEUR wrote:
Sorry, I do not understand everything. Can you detail please?
2013/6/17 David | StyleFlare <da...@styleflare.com
<mailto:da...@styleflare.com>>
You will have to do IP auth on the Kamailio coming from Asterisk
On 6/17/13 4:07 PM, Mickael MONSIEUR wrote:
Hello,
First, i have 7 years experience with Asterisk, but I started a
project with Kamailio, forgive me in advance if I say silly
things...! ;-)
I set up a classic Asterisk / Kamailio configuration:
sip phones -> kamailio -> asterisk -> sip trunks/pstn.
When a call comes from the PSTN side, if I configure Asterisk as
follows:
[012345678]
type = friend
username = 012345678
secret = xxxxxxx
host = dynamic
fromdomain = sip.mydomain.com <http://sip.mydomain.com>
fromuser = 012345678
Standard mode:
exten => 012345678, 1, Dial(SIP/012345678) -> The call is
redirected on the phone by Kamailio ! :-)
------------------------------------------------------------------------------------------------------------------------------------------------
Trunk mode:
[mytrunk]
type = friend
username = mytrunkUser
secret = xxxxxxx
host = dynamic
fromdomain = sip.mydomain.com <http://sip.mydomain.com>
fromuser = mytrunkUser
exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is
rejected by Kamailio....
exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is
rejected by Kamailio ....
My question is how to allow the routing of multiple numbers
(trunk mode) in a SIP account with Kamailio?
Best regards,
Mickael
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