[OpenSIPS-Users] Opensips + asterisk 1.4
Nick Khamis
symack at gmail.com
Thu Jul 18 23:05:02 CEST 2013
There is also:
1. modparam("auth_db|usrloc|uri", "use_domain", 1)
Please change that to 0. It's been a while since I have dealt with REGISTER
authentiacation issues. Are you sure
you need it? It's quite a resourceful process as the number of clients
increase. What we do now, is use:
1) The address table
2) Dialplan
3) Dynamic Routing
4) IPTables
To enforce who's INVITE gets processed by our servers. No registration
required.
If you really want to handle REGISTER, I will take a closer look. Until
then maybe look at Chapter 5 (Page 90),
of
ftp://115.146.120.141/voIP/Building%20Telephony%20Systems%20with%20OpenSIPS%201.6.pdf.
I know the
answer is in there because I dealt with your issue a long time ago.
Kind Regards,
Nick
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