El Miércoles, 24 de Junio de 2009, Steven E. Ames escribió: > Hi. I have an OpenSIPS and Asterisk setup. Incoming calls come from VoIP > Carrier to OpenSIPS. OpenSIPS does some dbaliases because sometimes > multiple numbers are assigned to same asterisk extension. This all works > great. > > However, when asterisk makes a call outward for a number that is actually > local then I get a Loop and asterisk is unhappy. > > Example: I have 2 incoming DIDs (111-222-3333 and 111-222-3334). On > OpenSIPS in dbaliases I translated 111-222-3333 to 111-222-3334 and send it > to asterisk. All is fine. On asterisk it knows about 3334 but not 3333. So > if another extension on asterisk dials 111-222-3333 it gets to OpenSIPS. > OpenSIPS does know about 3333 and knows how to handle it. It converts it to > 3334 and sends it back to asterisk. Voila. Loop. Now the actual behavior is > what I want but I want to modify the SIP INVITE such that asterisk will > accept it and not gripe about the loop. > > Any pointers?
This is a kwnown bug of Asterisk. -- Iñaki Baz Castillo <[email protected]> _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
