Help with Asterisk config - no audio external incoming calls

[pcdunlop]

Hi,

I am encountering one issue with my new Asterisk 1.8 local server with FreePBX and I was hoping for some advices.

There is no audio on external calling calls (both ways). I mean external any phone other than my other freephonline number (setup directly on linksysPAP2 ATA, in the same network). So If receive an incoming call from my other FPL number it works great, but any other external phone calling in there is no audio both ways - the call gets connected though. Any outgoing calls works fine.

I googled and it could be a NAT network issue. Since I am behind a NAT, I have nat=yes in the peer details and NAT enabled in the Asterisk SIP Settings. I have also put the external IP in the Asterisk SIP Setting but that didn't solve the problem. What's it weird is that if I setup "Local Networks", I don't get audio from calls within my network as well! So I just left External IP populated (IP Configuration = Static).

I see a lot of "Found RTP audio" in the call that audio works and just a few for the other call. I also see "Found audio description format G729a" only in the working call. Both are using the same Incoming route (which forward calls to my cell phone via ring group) and the same Freephoneline route. My router is Linksys WRT54G and ISP Teksavvy cable.

Any ideas?

Thanks!

[Jake]

Could be a codec problem. What is on the end of extension 101? (type of ATA, softphone, etc)

Also, could you post your PEER details (take out your secret and username)

One last thing, in the details for that extension, have you got anything in the allow or disallow boxes in the Device Options section?

[pcdunlop]

Jake, thanks for your response . I spent more time and I think it is fixed now. I had to enable "DMZ Host IP Address" in the router (to the IP where asterisk is running) and I was able to use the External IP and the Local Networks (192.168.1.0/255.255.255.0).

My peer details is as follows (just to make sure it's matching yours):

context=from-trunk
host=voip.freephoneline.ca
qualify=yes
dtmfmode=rfc2833
username=1289XXXXXXX
secret=SECRET
type=friend
insecure=invite&port,invite
disallow=all
allow=G729&ulaw
trustrpid=yes
sendrpid=yes
nat=yes
canreinvite=nonat

The end is my Telus cell phone which I configured with Incoming route + Ring Group for call forwarding. I don't have extension configured, only trunk, 2 Incoming routes (one with DISA script and the other to forward the calls in) and 1 outgoing route.

[Jake]

I would strongly advise you take it out of DMZ, especially as you put your IP in the post before.

I sent you a PM

If this fixed your problem, then it sounds like you need to forward some ports. DMZ is a bad idea, good for testing, but bad overall.

[pcdunlop]

Ops! I reverted the changes and used port forwarding instead, it seems to be working. Thanks again.