18 Jun
2014
18 Jun
'14
11:42 a.m.
Hi,
I have a carrier who requires SIP authentication on calls, can anyone point
me in the right direction? Normally in Asterisk or Freeswitch you can
authenticate a peer, but can't see how to do this in Kamailio.
Thanks
Keith
18 Jun
18 Jun
12:06 p.m.
New subject: SIP Trunk
On Wednesday 18 June 2014 10:42:02 Keith wrote:
I have a carrier who requires SIP authentication on
calls, can anyone point
me in the right direction? Normally in Asterisk or Freeswitch you can
authenticate a peer, but can't see how to do this in Kamailio.
You need the uac module and handle the 401 responses
http://kamailio.org/docs/modules/4.0.x/modules/uac.html#idp1684080
For example if you have dispatcher that need authentication for invites (where
RTF_DISPATCH is the dispatcher its failure route):
modparam("uac","auth_realm_avp","$avp(s:uac_auth_realm)")
modparam("uac","auth_username_avp","$avp(s:uac_auth_user)")
modparam("uac","auth_password_avp","$avp(s:uac_auth_pass)")
failure_route[RTF_DISPATCH]
{
if (t_check_status("401")) {
$avp(uac_auth_realm)=$td;
$avp(uac_auth_user)="username";
$avp(uac_auth_pass)="password";
uac_auth();
t_relay();
exit;
}
...
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participants (2)
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Daniel Tryba -
Keith