2009/5/22 John Quick <[email protected]>: > James > > The default behaviour for Asterisk is to send re-invites to the connected > parties that will re-direct > the RTP stream to go directly between the end-points instead of going through > Asterisk. > > In theory the option "canreinvite=no" should prevent this happening, but I > have never found it works.
Perhaps "canreinivite=never" will force it definitively (not sure). > Instead, the trick that always works for me is to add an option to the Dial > command that tells > Asterisk to look for DTMF during the call. Suitable options include "t", T", > "h", "H", "w", "W" or > "L". "t" and "T" options will force RTP through Asterisk if the peers are configured with dtmfmode=rfc2833. If you set "dtmfmode=info" then the audio will not force to pass through Asterisk (it will depend on other factors as canreinvite and so). > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial > > ...and this means it is really a question related to Asterisk and not > OpenSIPS. For sure :) Unfortunatelly it seems that people integrating OpenSIPS with Asterisk always comes to OpenSIPS maillist to ask question, in fact, about Asterisk :( -- Iñaki Baz Castillo <[email protected]> _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
