The var is now begins rtp_ rather than sip_ this applies to all media related vars. On Feb 6, 2014 12:31 PM, "Yuriy Nasida" <nasida@live.ru> wrote:
Hi there!
I know that it should be simple but when i add
<action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/>
nothing happens...
I didn't see fmtp:18 annexb=no in SDR of B-led
the callflow:
Skype --> FS ----G729a---> VOIP
Please advice Thanks.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org