hi,
I use asterisk 1.8.xx version and I take no samples for alawtolin warning everytime , I faced one way audio problems and call drop. nowadays calls dropped after 1 second.
this is codecs:
speex - - - - - - - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - - - - - - - -
g726 6999 3000 2001 2001 4000 2001 2000 3000 4999 - - - 2001 - - 3001 - - 2001
g722 5000 1001 2 2 2001 2 1 1001 3000 - - 3001 - - - 1000 - - 2
siren7 - - - - - - - - - - - - - - - - - - -
siren14 - - - - - - - - - - - - - - - - - - -
slin16 6000 2001 1002 1002 3001 1002 1001 2001 4000 - - 4001 1000 - - - - - 1002
g719 - - - - - - - - - - - - - - - - - - -
speex16 - - - - - - - - - - - - - - - - - - -
testlaw 5000 1001 2 2 2001 2 1 1001 3000 - - 3001 2 - - 1002 - - -
from cli :
[2018-02-12 19:40:24] WARNING[30189]: translate.c:206 framein: no samples for alawtolin
[2018-02-12 19:37:14] WARNING[25160]: chan_sip.c:3822 retrans_pkt: Retransmission timeout reached on transmission xxx@x.x.x.x for seqno 131 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
any idea?