One way audio

Asterisk Asterisk Support
1

First time setting up asterisk and I can now make incoming and out going calls but with 1 way audio.

The person making the call can send audio and the person receiving the call can hear the audio, but the person making the call cannot hear or receive any audio.

so if I make a call from a landline to asterisk I can hear the audio fine on my PC from the landline but I cannot talk back to the landline

and

If I make a call from my PC to a landline I can hear what is been said on the PC but cannot talk back to the PC

I am using sipgate.co.uk for the voip calls.

hear is the relevant info from my sip.conf

[general]

port = 5060
bindaddr=0.0.0.0
externip=77.86.68.110
localnet=10.20.0.0/255.255.0.0
disallow=all
allow=alaw
allow=ulaw

register => 1203144:passwd@sipgate.co.uk/1203144

[sipgate]
type = friend
secret = passwd
;insecure = invite
username = 1203144
defaultuser = 1203144
fromuser = 1203144
context = inbound
fromdomain = sipgate.co.uk
host = sipgate.co.uk
outboundproxy = proxy.live.sipgate.co.uk
qualify = yes
dtmfmode = rfc2833
externip = 77.86.68.110
nat = yes

[sipgate-out]
type=friend
secret=passwd
host=sipgate.co.uk
fromdomain=sipgate.co.uk
qualify=no
username = 1203144
defaultuser = 1203144
fromuser = 1203144
dtmfmode=info
insecure=invite
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
nat=yes

[mike]
type=friend
username=mike
secret=mike
host=10.20.30.1
context=users
mailbox=100@default
fromuser=mike

and my extensions.conf

[users]
include => sipgate
include => phones

[sipgate]
exten => _7.,1,Dial(SIP/${EXTEN:1}@sipgate-out,60,tT)
exten => _7.,2,Congestion
exten => _7.,3,Busy
exten => _7.,4,Hangup


[inbound]
exten => 1203144,1,Answer
exten => 1203144,2,Dial(Local/#100@phones)
exten => 1203144,3,Hangup()

[phones]
exten => _#100,1,Dial(sip/mike)

I have been googleing this for ages and tryed loads of differant settings but nothing seems to work,.
I have forwarded port 5060 from my router to the * box and the same for ports 10000-20000. Audio can pass both ways but not on the same call so I dont think its a nat issue

any help will be much appreciated

Mike

2

You need to extract the SDP negotiation from the SIP to have a chance of working out what is wrong, but I would think your router is the problem area.