Hello,
I am trying to configure an inbound/outbound sip trunk in Asterisk.
The issue is with the outbound, as soon as I try to place an outbound call I receive a congestion tone however the destination is receiving the call without voice due to caller is disconnected
– SIP/trunk1-0000000d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
However everything is working fine for inbound.
I noticed by changing qualify from yes to no it works fine.
Do you know why it is not working with qualitfy to yes ?
My Asterisk release is 1.8
language=be
context=default
srvlookup=yes
udpbindaddr=0.0.0.0
allowoverlap=no
rtcachefriends=yes
disallow=all
allow=alaw
register => user:password@sip.nomado.eu:5060
[trunk1]
type=peer
defaultuser=user
secret=password
fromuser=user
fromdomain=sip.nomado.eu
host=sip.nomado.eu
insecure=invite
nat=yes
port=5060
canreinvite=no
disallow=all
allow=alaw
qualify=no
context=trunk-in