[OpenSIPS-Users] Integration with Asterisk/Trixbox

[Iñaki Baz Castillo]

2009/5/22 John Quick <John.Quick at smartvox.co.uk>:
> 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
<ibc at aliax.net>