2014-09-28 14:01 GMT+08:00 Markus <unive...@truemetal.org>: > Am 27.09.2014 17:28, schrieb d tbsky: >> >> can someone give an example for the function? thanks for the help. > > > Not a programmer here, just grep -r'ed through the code, but maybe try one > of these: > > G711A > G711_ALAW
thanks a lot for help!! I tried both but none works. maybe this function can not work like the old channel variable "SIP_CODEC", which can change inbound call codec. but I do notice something different between chan_sip and chan_pjsip. I use zoiper softphone for testing: when I dialout sip trunk with chan_sip, the remote peer rings, and zoiper now shows what codec to use. if I use "SIP_CODEC" before dial to change the codec, zoiper will use the new CODEC, but asterisk internal won't change and still transcoding in the middle.(at least "core show channel sip/xxxxx" told me transcoding) when I dialout sip trunk with chan_pjsip, the remote peer rings, but zoiper didn't show what codec to use. only after the callee answer the phone, zoiper shows what codec to use. so it seems chan_pjsip have better chance to do the right thing without transcoding. it's sad that chan_pjsip won't select best codec match two peers automatically without transcoding. but I hope it at least can provide a magic function or channel variable like "SIP_CODEC/SIP_CODEC_INBOUND" to make correct codec selection. Regards, tbskyd -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users