[asterisk-dev] Jitter Buffer problem

[Ed]

Igor A. Goncharovsky wrote:
> I have found problem with one of my connected SIP trunk. Recently I have
> enable jitter buffer on my test asterisk and get many log messages on
> outgoing SIP call like this:
>
> [Mar  5 15:11:48] WARNING[24195]: abstract_jb.c:318 ast_jb_put:
> SIP/101-b4506410 received frame with invalid timing info:
> has_timing_info=0, len=0, ts=0, src=lintoalaw
>   


you use ipp-based g.729 codec? try lastest from http://asterisk.hosting.lv/

ps: i think you select wrong list for question