I need help. I got one way audio with one specific provider and found something in tcpdump log.
SIP client s.s.s.s to Asterisk a.a.a.a - Provider p.p.p.p - PSTN
After connection has been created, one package received from provider to tell asterisk another RTP ip to connect, so the connection changed to
SIP client s.s.s.s to Asterisk a.a.a.a - Provider o.o.o.o, then I see connection to be,
s.s.s.s:s => a.a.a.a:a
a.a.a.a:a => o.o.o.o:o
p.p.p.p:p => a.a.a.a:a
( No connection o.o.o.o:o => a.a.a.a:a or a.a.a.a:a => p.p.p.p:p have been found after rtp ip change)
result is I (SIP client) can hear other party, but other party can’t hear me. Is there anything I should setup in asterisk for rtp ip change in sip. I use 1.6 and 1.8, same issue. Please help.