Matt, Asterisk version == 1.4.22 dtmfmode == info calls are bridged through Asterisk (canreinvite=no)
Jonathan On Sun, Dec 28, 2008 at 3:23 PM, Matt Florell <astma...@gmail.com> wrote: > On 12/28/08, jonathan augenstine <jaugenst...@gmail.com> wrote: > > I am trying to resolve an issue and I believe it is my configuration. > The > > scenario is that I have a SIP detected on the server. The dial plan then > > makes a local connection to another part of the dial plan. The new dial > > plan extension then places another SIP call out to a SIP phone. When the > > call is accepted there is streamed from the calling SIP phone. When the > > audio is complete a DTMF is transmitted to Asterisk. The DTMF is > detected > > by Asterisk but it does not get passed through to the other SIP phone. I > > would like the DTMF to pass-through to the other SIP phone. Is this a > > configuration issue? Or do I need to handle this on the dial plan level? > > > > Jonathan > > Asterisk version? > > What are both dtmfmodes set to for each SIP endpoint? > > Are the calls natively bridged or bridged through Asterisk? > > MATT--- > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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