I am trying to dial through my asterisk machine from phone A to phone B.
My DID is registered properly with the SIP provider. When I dial from
A to B it looks fine so far.
A rings B and B can pick up and the call is bridged. However, I don’t
hear any audio so therefor it is not working.
I am running Asterisk 1.8 on a cloud server. I have had the same
configuration running on a physical machine with a similar
Thoughts? I know I posted this yesterday but was hoping for some more
Zip*CLI> sip show registry
Host dnsmgr Username Refresh
sip.callwithus.com:5060 N xxxx 105
Registered Tue, 07 Dec
1 SIP registrations.
canreinvite=no ;if your asterisk box is behind a NAT ro
;register => xxxx:firstname.lastname@example.org
register => xxxx:email@example.com
CONSOLE=Console/dsp ; Console interface for demo
exten => s,1,Answer()
exten => s,n,Dial(SIP/callwithus/12222222222)
exten => s,n,Wait(2)
exten => s,n,Hangup()