ANyone ever ran into a situation when Call coming from Call Manager into asterisk, is successful coming across – but no Audio ???But then the next call – audio is heard – its once in a great while no audio – most time it works.Anything I might look ..

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we are migrating from chan_sip to pjsip

i want logs like this about pjsip endpoints

[Oct 14 17:20:36] NOTICE[35629] chan_sip.c: Peer endpoint22 is now Reachable. (15ms / 2000ms)

is it possible?

thanks

..

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