I have configured asterisk with flowroute. The registration is successful and outbound call connects but i am unable to hear any sound. The error i see in logs is
Failed to get 160 samples from write factory 0x7fd93000c2c8
[2017-04-13 12:10:20.605] DEBUG[19683][C-00000000]: audiohook.c:263 audiohook_read_frame_both: Read factory 0x7fd93000b688 and write factory 0x7fd93000c2c8 both fail to provide 160 samples
[2017-04-13 12:10:20.625] DEBUG[19683][C-00000000]: audiohook.c:316 audiohook_read_frame_both: Failed to get 160 samples from write factory 0x7fd93000c2c8
[2017-04-13 12:10:20.625] DEBUG[19683][C-00000000]: audiohook.c:263 audiohook_read_frame_both: Read factory 0x7fd93000b688 and write factory 0x7fd93000c2c8 both fail to provide 160 samples
[2017-04-13 12:10:20.645] DEBUG[19683][C-00000000]: audiohook.c:316 audiohook_read_frame_both: Failed to get 160 samples from write factory 0x7fd93000c2c8
My configurations are as follow
sip.conf
[general]
register => username:password@sip.flowroute.com
allow=ulaw
sip-custom-contexts.conf
[flowroute] ; keep this lowercase, do not change format
type=friend
secret=username
username=password
host=sip.flowroute.com
type=peer
transport=udp
port=5060
dtmfmode=rfc2833
context=to-pstn ; change to 'ext-did' or 'from-trunk' for asterisk@home
canreinvite=no
allow=ulaw
;allow=g729 ;uncomment this line if you have G.729 licenses installed.
insecure=port,invite
fromdomain=sip.flowroute.com
keepalive=yes
qualify=no
progressinband=yes
rtp.conf
[general]
rtpstart=10000
rtpend=2000
Kindly help in this regard