On Sunday 26 April 2009, Esben Stien wrote: > Nicola Mfb <nicola....@gmail.com> writes: > > a dialer that interacts both with FSO and the VoIP stack. > > We should use freeswitch as the central daemon for voice and video > sessions. This should talk to the GSM daemon. A dialer should just be > a simple SIP client. This means we can use any SIP client out there to > make the call and it won't matter how this call is setup in > freeswitch.
Except that because of the audio routing required it won't be that easy. You will have to use the one channel (say left) for connecting PCM to the GSM, and the other for PCM to the mic/earpiece. I doubt generic SIP clients and freeswitch, or asterisk's chan_alsa, will play nicely under those conditions. _______________________________________________ Smartphones-userland mailing list Smartphones-userland@linuxtogo.org http://lists.linuxtogo.org/cgi-bin/mailman/listinfo/smartphones-userland
