-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 >You do need to have this enabled in the dialplan dial strings to >enable transfers. u should use something like this:
[from-sip] exten => 101,1,Dial(SIP/sip1,20,tTr) from http://www.voip-info.org/wiki-Asterisk+cmd+Dial: The options parameter, which is optional, is a string containging zero or more of the following flags and paramters: t: Allow the called user to transfer the call T: Allow the calling user to transfer the call r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBClf24Q/49nIJTlwRAg/kAJ90/tEQZXEIVe+A1WTM7HDtQGF1dgCeLrCG 01E4lkdvIbjpGrvMoiGu324= =lgdl -----END PGP SIGNATURE----- _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
