Connecting from SipBroker, no audio
 

I am a newbie here, so please be patient!

I am running Asterisk behind a NAT with dynamic DNS. I want to use Voxalot with the Sip Broker PSTN access numbers, so that friends interstate can dial up Sip Broker, put in *061 and my Voxalot number, and connect to me.

Well, testing it from my PSTN phone, the thing works in that my VoIP phone rings, but there is no audio in either direction. I have tried turning off the firewall, but it makes no difference. I have tried forwarding ports 5060 and 10,000 - 20,000, but that doesn't help either.:mad:

I have disabled all codecs except ulaw, but that makes no difference.

Any suggestions please?:confused:

Further
 

I should have added that these are my (relevant) settings in sip.conf, and that Asterisk is registering OK with Voxalot... (BTW, I have an a/c with iiNet, and their VoIP works perfectly on the same Asterisk server.)

[general]
register => 830***:*secret*@voxalot.com/830***

[voxalot]
context=special
type=peer
disallow=all
allow=ulaw
authuser=830***
fromdomain=voxalot.com
fromuser=830***
host=voxalot.com
insecure=very
qualify=yes
secret=********
username=830***
canreinvite=no
nat=yes
;
;
[vox-in]
;
type=user
host=voxalot.com
disallow=all
allow=ulaw
context=incoming
fromuser=830***
insecure=very
secret=*******
username=830***
dtmf=rfc2833

try forwarding to sip:600voxalot.com for echo test and see if you can hear yourself. Other suggestions are to try again default codecs and with and without stun. Which ATA are you using?

" ...try forwarding to sip:600voxalot.com for echo test and see if you can hear yourself...."

Yes, if I call forward to the echo test - or even the echotest on blueface.ie - I can hear myself: it's fine. (That is, I'm dialling in from PSTN to SipBroker, putting in the code for Voxalot and call forwarding from Voxalot to the ech test, and that's fine.) The ATA is a Sipura 3102, but I get the same problem if I use a softphone on my laptop, so it can't be the ATA... has to be a problem with something like the NAT, the ports or even the DNS, I think. BTW, I had much the same problem with Faktortel, but iiNet and bbpglobal are fine.

- Martin

A couple of things to try. In [general] in sip.conf add:

externip=<your dyndns hostname>
localnet=192.168.0.0/255.255.255.0 <== assuming your internal ip addresses are in the 192.168.0.* range.

Also if you are planning to use *061 change all host= fields to use voxalot.com.au and the register => field also. The from domain can remain voxalot.com

Thank you for your help. I'm using externhost= , so don't seem to need externip. Actually, I ran Sip debug and found out that the real problem was that I had mixed up 'peer' and 'user' definitions... (I always find this confusing!:( ) Asterisk was looking for the extension in the wrong context. Now that's fixed and all is well. I'm exstatic!:D Maybe it's the same problem with Faktortel!

Thanks for the help, and the service! - Martin