Hi
I had a sip trunk set-up to use voipstunt and everything has worked find for the past 3 months. For the past 2 days now the calls get connected and the called party can hear me but I cant hear them!
My network and non of my settings have been changed.
Any suggestions
a reboot of your Asterisk PC and router might fix the problem. otherwise you’ve given way too little information for anybody to know how to start helping you.
My router is connected to my ISA server. The ISA server, asterisk server and phones are connected via a hp procurve switch.
Router — ISA – Switch – Internal Network
Router - (10.0.1.1)
ISA - (10.0.1.2) & (10.0.0.11)
Asterisk - (10.0.0.55)
Phones - (10.0.0.100 - 250)
My sip settings are:
allow=ulaw&alaw
authuser=username
canreinvite=no
disallow=all
dtmfmode=inband
fromdomain=voipstunt.com
fromuser=username
host=sip.voipstunt.com
insecure=very
nat=yes
secret=password
type=peer
username=username
Everything seems fine except that i cannot hear callers when I dial out of my VOIPStunt trunk. If i dial out of other sip trunks i.e. sipgate everything works fine and we can both hear each other but not with voipstunt.
As I said it was working for months and suddenly stopped working - nothing seems to have changed on my end.
Any ideas whats wrong or do you need more info?
these sort of problems take a lot of time to debug. since your other sip trunks are working it’s probably not a router issue, but i’d reboot it just the same (and ISA). i suggest this because it has fixed this type of problem for me in the past.
turn on RTP debugging. do you see incoming RTP? probably not, just outgoing. check whether the router or ISA are blocking the incoming RTP by looking at the logs.
do you have a dynamic ip address?
you didn’t run out of your 120 ‘FREEDAYS’ on VoipSunt, did you?
Also, could it be possible that VOIPSTUNT has changed their RTP ports and now ISA is not letting them in?