No outgoing sound over outbound SIP trunk


#1

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?