Hello,
I have an Asterisk machine with 2 interfaces, one doing NAT to the inside LAN and the other with a routable public IP.
However, when I do I dialplanning routing to the SIP provider VoipBuster the called party can’t hear me despite me hearing them perfectly (sometimes with and a slowed down robotronic voice).
These configurations have worked in the past but in the past two weeks they’ve misteriously stopped functioning and since then i’ve been tearing off my hair!
I route an inside call to the provider using a Dial(SIP/${EXTEN}@voipbuster) and paste below the contents f my sip.conf:
[general]
useragent=“IPB”
context=SER_Asterisk
bindport = 5090 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
srvlookup=yes
disallow=all
allow=gsm
allow=ilbc
allow=speex
allow=lpc10
allow=ulaw
allow=alaw
autocreatepeer=yes
[ser]
type=friend
username=asterisk
secret=asterisk
host=192.168.69.254
port=5060
insecure=very
[voipbuster]
type=peer
host=sip1.voipbuster.com
username=
fromuser=
secret=
I’ve been able to have the trunk working again by changing the [voipbuster] entity to the following config.:
[voipbuster]
type=peer
host=sip1.voipbuster.com
username=
secret=
fromdomain=sip1.voipbuster.com
fromuser=
dtmfmode=inband
canreinvite=no
Any hints?