I think the file should be call config_site.h Also send the SDP use in the INVITE to check it as well Sent from my iPad On Jun 5, 2010, at 4:21 AM, Shrouk Khan <shroukkhan at softverk.is> wrote: > Hi, > i have successfully compiled the pjsip(V1.6) with APS support and have been testing the symbian_ua_gui on my nokia E66 with my asterisk server . But it seems that the calls are always made in g711 . > codec. I have tried to force the g729 in my config_site_sample.h header file in the following manner: > > #ifdef PJ_CONFIG_NOKIA_APS_DIRECT > > /* MUST use switchboard rather than the conference bridge */ > #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 > > /* Enable APS sound device backend and disable MDA & VAS */ > #define PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA 0 > #define PJMEDIA_AUDIO_DEV_HAS_SYMB_APS 1 > #define PJMEDIA_AUDIO_DEV_HAS_SYMB_VAS 0 > > /* Enable passthrough codec framework */ > #define PJMEDIA_HAS_PASSTHROUGH_CODECS 1 > > /* And selectively enable which codecs are supported by the handset */ > #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMU 0 > #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA 0 > #define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR 0 > #define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 1 > #define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC 0 > > #endif > > > as you can see i have disabled all the passthrough codecs except g729. but still the g711 happens . > > do i need to do something else for that? i would like to contribute to an wiki on how to enable g729 on symbian pjsip > -- > Regards > > Shrouk Khan (Khan) > System Administrator / Telecommunication System Developer > Office: +354 4400807 (Reykjavik) > +44 2031370800 (London) > Mobile: +66 875049439 (Bangkok) > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
