asterisk users - Dec 2004 - One-way audio with SIP client only on certain calls

Hello.

I have an * server set up on a public IP. I have SIP clients at three
different locations, all behind NATs. I have all the SIP users set up
this way:

[user1]
type=friend
username=user1
secret=password1
callerid="User 1"<101>
host=dynamic
qualify=yes
context=outgoing

All three SIP clients are configured to use STUN (using
stun.fwdnet.net:3478).

Furthermore, I have LiveVOIP set up like this for incoming/outgoing
calls:

[livevoip]
type=friend
username=peer1
fromuser=peer1
secret=password
host=livevoip.ip
context=from-livevoip

Calls between location 1 (using a Sipura SPA-3000) and location 2 (using
a Grandstream Budgetone 101) go through perfectly. Calls from the PSTN
(through LiveVOIP) to location 1 or location 2 go through perfectly as
well.

Now, I recently added a client at a third location (also using a
Grandstream Budgetone 101, same settings as location 2). This is what
works and what doesn't:

1. calls to this new client *FROM* location 1 or 2 result in a good
call.
2. calls from this new client *TO* location 1 or 2 result in one-way
audio from the new client to the callee.
3. calls to this new client *FROM* LiveVOIP result in one-way audio from
the new client to the caller.
4. calls from this new client *TO* LiveVOIP result in a good call.

The addition of canreinvite=no to the SIP entry for the client at
location 3 *AND* placing the SIP device in the DMZ of the router at that
location, solved #2. Doing only one or the other did not solve the
problem on it's own.

However, no matter what I try, I cannot figure out #3. I tried adding a
notransfer=yes to the IAX entry for LiveVOIP, but that did not solve the
problem.

Bear in mind that the users at location 1 and 2, both behind NATs and
one using the exact same devices, are able to make/receive all four
categories of calls successfully.

I investigated a little, and it seems the IP that the SIP device
registers with the * server is not the IP of their home router (a
Linksys something-or-the-other), but going to http://ip.address/ gives
me a page with the heading "MicroTik RouterOS". However, I believe
their
home router has a "real" IP, not a private IP. So, for some reason,
STUN
(or something else) is seeing their IP as their ISP's router. It almost
seems like their connection is double-NATted?

I am unsure what steps to take next, so any help would be greatly
appreciated.

--
Nabeel Jafferali
tel: 647.722.8457 x201
     718.606.4190 x201
fwd: 46990 x201
email/msn: nabeel<at>jafferali.net

----- Original Message -----
From: "Nabeel Jafferali" <nabeel@jafferali.net>
To: <asterisk-users@lists.digium.com>
Sent: Saturday, December 18, 2004 5:51 PM
Subject: [Asterisk-Users] One-way audio with SIP client only on certain
calls


Hello.

I have an * server set up on a public IP. I have SIP clients at three
different locations, all behind NATs. I have all the SIP users set up



Try this:

[user1]
type=friend
username=user1
secret=password1
callerid="User 1"<101>
host=dynamic
qualify=yes
context=outgoing
canreinvite=no