ASA5506 cuts RTP during local calls - how to fix it?

Question

SIP

There is an ASA5506, a dozen Panasonic KX-HDV100 phones, a Beeline provider and an Internet channel for telephony. Phones are set up for one Beeline account and 10 virtual lines, each line. When calling from these phones to the outside, everything works fine, all traffic goes, the voice is in order and so on. When calling from one of the phones to the other, all SIP traffic passes perfectly, and the voice from both phones of the tsiska cuts when it is sent outside. I collected the dumps, in the dumps all packets comply with the SIP standard, i.e. Beeline in the SDP packet sends an IP:port on its device, the phone sends RTP there, but the ace drops it. Where to dig? inspect sip is included in default inspection, without it the beeline sends 403 and not a single phone works. Otherwise, the config is standard, except for open udp permission from outside to inside on ports from the RTP range:
access-group outside2_access_in in interface outside2
access-list outside2_access_in extended permit object rtp-local any4 192.168.103.0 255.255.255.0 object
service rtp-local
service udp destination range 10000 20001
local, and in fact there is no difference in SIP / SDP packets. Well, I don't see her at all.

Answer 1

try troubleshooting with packet-tracer:
asa-5540s# packet-tracer input ?