[Asterisk-Users] SIP conections, with RTP not going trough Asterisk

stoffell stoffell at gmail.com
Mon Apr 17 05:04:19 MST 2006


On 4/17/06, Alex Mosburger <alex.mosburger at distanzfotos.at> wrote:
> -) * needs to listen to DTMF tones during the call (for transfers or any
> other features)

Does this mean you cannot do any blind or attended transfer? or only
the # transfer option (asterisk built-in, from features.conf) doesn't
work?

cheers



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