[ygator]

I have Asterisk running internally on my home network. I am natted with one public IP address. I have my router set to DMZ my nslu2 which is running Asterisk. On my internal LAN everything works great. I am now trying to get a PAP2 on the outside world to connect to asterisk. The Pap2 is also on a natted network. See configurations below.

I can make and receive calls with this setup, however:
1) I only hear audio from asterisk
2) Asterisk does not know when I hang up

I have an extension that says the time and hangs up.
I can call that extension and hear the time.
If I hang up before the time is done I see the debug output of asterisk showing that I did not hangup and that it is still saying the time and then hangs up.

What am I missing to get this to work totally?

Thanks

The Pap2 is setup as follows:
NAT Mapping Enable = Yes
NAT Keep Alive Enable = Yes
Use Outbound Proxy = Yes
Outbound Proxy = fwdnat2.pulver.com:5060

The entry in SIP.conf in asterisk for the Pap2 is:
type=friend
context=default
host=dynamic
nat=yes
canreinvite=yes

[mberlant]

There are several things you mention that are contrary to the recommendations of the Asterisk developers as well as the advice previously rendered here in this forum. You may want to read through this forum before progressing further.

What made you decide to use FWD's Outbound Proxy (which forces all RTP packets through that proxy) to talk between your own PAP2 and your own Asterisk?

What made you set canreinvite=yes when even the comments in sip.conf.sample specify that you must set canreinvite=no when you set nat=yes?

Why is the sip.conf entry for the PAP2 so sparse?

How did you configure your PAP2? Did you use the Voxilla Wizard (for FWD w/STUN, as I have recommended)? How did you configure sip.conf? Did you use the advice mentioned here in this forum from other members who have succeeded?

The one good thing is that, with Asterisk running on NSLU2, you are not constrained by the limitations of AMP.

[ygator]

Okay I got it working. I had forgot to put my ip address in the externip.
No outbound proxy or stun is being used.

I'll go ahead and send my responses to your reply since I had already answered before I noticed the externip.

Originally Posted by mberlant

There are several things you mention that are contrary to the recommendations of the Asterisk developers as well as the advice previously rendered here in this forum. You may want to read through this forum before progressing further.

I've been reading the Asterisk Wiki. I looked through here, but I did not see anything that could help me. I look again.

Originally Posted by mberlant

What made you decide to use FWD's Outbound Proxy (which forces all RTP packets through that proxy) to talk between your own PAP2 and your own Asterisk?

The solution by Mervyn Sirju located here
http://www.voip-info.org/wiki/index....+NAT+solutions

I was using FWD, but with the bug in the SPA3000 with CID I was unable to know who was calling from the PSTN. I found that asterisk did not care about the SPA3000 bug so I set off to using it. Internally is working fine. I am now trying to allow outside clients get connected. Also I'd actually like to not rely on anything else if that is possible.

I originally had no outbound proxy, but could not get that to work. I then tried FWD stun server, but had no audio at all. I then did a DMZ to the PAP2 on the other network and I had one way audio. I then saw the outbound proxy thing in the wiki and tried that.

The closest I have come to making it work is one way audio, but asterisk does not know when I hang up.

Originally Posted by mberlant

What made you set canreinvite=yes when even the comments in sip.conf.sample specify that you must set canreinvite=no when you set nat=yes?

I forgot to change it to no and have not gotten a chance to see if that is what is still causing my problem.
When the Pap2 was connected internal on my network I was trying to get two sip clients to talk to one another without going through the nslu2. I could not get this to work as I always saw the the network light flash on the nslu2 when the call was taking place.

Originally Posted by mberlant

Why is the sip.conf entry for the PAP2 so sparse?

I just showed what I thought was relevant. What is missing? Here is the entire entry:

[ext401]
type=friend
regexten=401
username=user401
secret=password
context=default
callerid="test" <401>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
mailbox=401@default

Originally Posted by mberlant

How did you configure your PAP2? Did you use the Voxilla Wizard (for FWD w/STUN, as I have recommended)?

The Pap2 was originally working on FWD. I just changed to point to Asterisk.

Originally Posted by mberlant

How did you configure sip.conf? Did you use the advice mentioned here in this forum from other members who have succeeded?

I modified the entry that was working internally. I have been using the Asterisk Wiki.

Originally Posted by mberlant

The one good thing is that, with Asterisk running on NSLU2, you are not constrained by the limitations of AMP.

I didn't know what AMP was, but I guess you are referrring to Asterisk Management Portal. I just edit the asterisk .conf files to change settings.