I have an Asterisk servers (recent SVN version 1.2) and two Sipura ATAs (one 2000 and one 1001).
I have Three-way Conf Serv and Three-way Call Serv enabled on both ATAs. When I make a SIP call from phone 1 to phone 2 on my Asterisk box, it works fine, then when I press the hookflash on phone 1, the caller on phone 2 is placed on hold (Asterisk MoH plays). I get the "second dial tone" on the Sipura and dial another extension (phone 3) from phone 1 and it rings and I am connected. Now, if I press hookflash again on phone 1, it switches the connection to phone 2, but phone 3 is placed on hold rather than getting conferenced in. Anyone got any idea why the three-way call might not be working in these circumstances? The message type for setting of Hookflash (none, AVT, INFO) seems to make no difference. I normally set it to "none" as it should be handled locally rather than get sent to Asterisk. Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
