[asterisk-users] Can you verify this bug?

[Barton Fisher]

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
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