Has anyone made this work? For me everything is fine until I switch
canreinvite form no to yes. What happens is that asterisk hangs on
"attempting native bridge" ... from what I understand "attempting
native
bridge" means that the RTP is routed through asterisk (just without any
codec translation) But it shouldn't do that ... right? ... canreinvite is
set to yes ...
What's the best way to deal with this issue? I've also read that the
only
way to get the following situation ...
UA --- NAT --- Internet --- NAT --- UA
... to work without passing the media path through asterisk is to use SER
together with asterisk. Is that still true or was that because I was
reading stuff from back in 2003?
Some other discussions mention that canreinvite will simply not work with
certain UAs .. is PAP2 one of those?
.. Couple of other discussions that I've seen conclude that passing media
stream UA-to-UA is just not practical when NAT is involved and is best to be
avoided all together ... I'd like to make it work because it seems like a
great way to save expensive server bandwidth. But if it will cause more
trouble than it's worth then I will probably pass the media path through
Asterisk and live with the fact that it will eat up my bandwidth.
Also, IAX is superior when dealing with NATs , does it also handle UA-to-UA
in NATed environment smoothly? What would be a good PAP2 alternative that
uses IAX?
This is my sip.conf:
[1001]
username=1001
type=friend
secret=****
qualify=yes
port=5060
nat=yes
mailbox=1001@default
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=yes
callerid="Test1" <1001>
... My PAP2 is configured with:
STUN=yes
STUN=stun.xten.net
NAT Keepalive = 15
Outbound proxy = blank
Proxy = IP of asterisk
Any suggestions?
Thank you,
Tomas
