-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 El domingo 24 de mayo del 2009 a las 19:38:30 -0300, Daniel Bareiro escribió:
> Now it would remain to find the cause of why I cannot call from a SIP > extension to an analog telephone. Perhaps it is by something related > to the contexts in the mentioned configuration files? I forgot to copy the output that I obtain in the CLI when I call to a SIP extension: [May 25 19:22:57] NOTICE[4813]: chan_sip.c:14721 handle_request_invite: Call from '201' to extension '1010' rejected because extension not found. Thanks for your reply. Regards, Daniel -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkobGyoACgkQZpa/GxTmHTcYdgCfW8RUyUY5e4pbxs5xC/9Fcpp7 58UAnRAfj2eUL8ZAtvgUxIwHvCv2OXDM =o+On -----END PGP SIGNATURE----- _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
