Hi, i installed asterisk 1.6.2 on a centos 5.5 os, I am working only with soft phones for it is a project. The problem is that i have connected to a sip provider but when i call in from an another account, it connects but no sound goes through. I’m not behind a nat, I was before but because of this i changed to a public IP. but the problems still do persist. here is my sip.conf settings:
[rynga]
type=friend
dtmfmode=rfc2833
context=ryngal
host=sip.rynga.com
username=*************
fromuser=*************
secret=*************
canreinvite=no
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=no
fromdomain=sip.rynga.com
insecure=invite
srvlookup=no
and my extensions.conf
exten => ***************,1,Progress()
exten => ***************,2,Playback(demo-echotest)
exten => ***************,3,Echo
exten => ***************,4,Playback(demo-echodone)
sorry for being so disorderly but i’m really new to linux and asterisks and this also is my first ever post. So please do point me in the right direction. Thanks