I'm stuck on 1.2 until I can pass DTMF from a SIP Trunk (Vitelity Virtual PRI) call towards a ZAP (TE410P using e&m wink) port. The call connects OK, I can hear DTMF with DNIS & ANI inband from asterisk to the external IVR, Voice is OK, but if any DTMF is required after the bridge has been made, they are muted. I posted on http://bugs.digium.com/view.php?id=12913 but I have got much notice. I was wondering if you could test this scenario to see if it in fact fails and post your results in bugs? Thanks, Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080702/be082b93/attachment.htm