On Jul 22, 2006, at 12:54 PM, Robert Jenkins wrote:
<snip>
On Jul 22, 2006, at 11:13 AM, Robert Jenkins wrote:
Oh well..
I already had localnet set:
localnet = 192.168.0.0 ; Internal NETWORK address
localmask = 255.255.255.0 ; Internal netmask
All the involved PCs & Sipura boxes are using 192.168.0.x addresses.
The Sipura boxes work, but the fact that asterisk is sending the
external IP to any device on the local network seems to me to be a
bug..
You didn't mention whether you were also forwarding ports
10000-20000 to the SIP Proxy (ie asterisk). Thats where the
actual RTP (voice
data) is passing. Also you need to be sure that there aren't
multiple clients on your lan all trying to use the same ports
for signaling (ie 5060), as this will fail.
Hope this helps.
Marty
The simple thing is that if I have 'externip' set, I can see on a
soft phone
(running on a PC on the same local subnet as asterisk) that it's
seeing a
call from another local device as coming from [EMAIL PROTECTED] -
which is
the external IP and as everything is inside the firewall there is
no audio
from the soft phone when the call answered.
If I comment out the 'externip' line & restart asterisk, the soft
phone then
correctly sees the local call as being from [EMAIL PROTECTED] and I
get
two-way speech.
Re. multiple clients using port 5060, I have seen comments both ways..
This is how I have it at present and it works (without externip, which
appears to be down to asterisk sending the wrong info & nothing to
do with
ports).
As has been said elsewhere, if online VoIP services with thousands of
connections work with a single port, why should there be a problem
smaller
numbers of clients?
They are exposed as a single IP address. A single port 5060 is fine
for your asterisk box. BUT if you expect calls from the outside of
your LAN to pass to SIP phones on the inside of your LAN, you need to
do one of two things. 1) Use separate ports for the softphones so
the NAT isn't confused, or 2) make sure canreinvite is set to no in
your extensions for the softphone.
If you don't do one of those two things, then what will happen is
that the caller from outside will connect to the softphone inside,
and then attempt to talk directly to the softphone. BUT since your
router is forwarding all port 5060 traffic to your asterisk box you
are no longer talking to each other.
You don't mention whether your test calls are coming from inside your
lan or outside? You aren't by chance running on a softphone on the
asterisk box directly?
Marty
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