Curious Problem With NAT
Hi list!
Since the internal calls work as expected and I can register my Asterisk on an external provider, I’d like to add a new feature and allow my mobile phone to connect to my Asterisk and manage calls.
Well, first of all, my Asterisk is NOT direct on Internet available, but behind a NAT. So I configured my sip.conf:
localnet2.168.200.0/24
externhost=myhost.noip.com externrefresh0
Then I added the peer in my users.con:
[00491771111111]
fullname = 00491771111111
secret = MYVERYSECRET
type=peer nat=yes qualify=yes qualifyfreq`
hassip = yes dahdichan = 1
transport=udp,tcp callwaiting = no context = default host = dynamic dtmfmode=rfc2833
dial=SIP/00491771111111
and finally “core reload”.
On my Gateway I configured the NAT so:
/sbin/iptables -t nat -A PREROUTING -p udp –sport 6060 -j DNAT –to-destination 192.168.200.120:5060
/sbin/iptables -t nat -A PREROUTING -p tcp –sport 6060 -j DNAT –to-destination 192.168.200.120:5060
/sbin/iptables -t nat -A PREROUTING -p udp –dport 6060 -j DNAT –to-destination 192.168.200.120:5060
/sbin/iptables -t nat -A PREROUTING -p tcp –dport 6060 -j DNAT –to-destination 192.168.200.120:5060
Well, the phone connect to the server and I can see it reachable:
OpenWrt*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status
00491771111111/0049177111 192.168.200.3 D N 40702 OK (1768 ms)
Well, now I call the mobile phone from another peer. It rings and I can answer the call. Wonderful!
But no word will be sent… 🙁
I cannot hear anything on my mobile phone and I cannot transmit a single word….
I tried to connect my mobile phone to a public VoIP-Provider and it works as expected, so I’m sure that the problem is on my network, but I can’t find it…
What am I doing wrong?
Thanks a lot Luca Bertoncello
(lucabert@lucabert.de)
11 thoughts on - Curious Problem With NAT
Have you tried NAT=force_rport ?
Ashwin
—–Original Message—
Ashwin Surendran schrieb:
No, not yet… I’ll try later and report to the list…
Have I to define (in Asterisk or Gateway) the ports?
Thanks Luca bertoncello
(lucabert@lucabert.de)
Ashwin Surendran schrieb:
OK, tried… I can transmit from my phone (aka: I hear my voice on another phone), but I’m not able to receive data (aka: I cannot hear what I say on the other phone).
Other suggestion?
Thanks Luca Bertoncello
(lucabert@lucabert.de)
What settings have you got for directmedia?
Could you try
nat=force_rport,comedia
directmedia=no
-Ashwin
—–Original Message—
Ashwin Surendran schrieb:
Tried. Peer always unreachable, call not possible… 🙁
Other idea?
Thanks Luca Bertoncello
(lucabert@lucabert.de)
Are you using the wifi on on the cellphone? The peer IP is showing as
192.168.200.3 which is not a routable address. Unless things have changed, double NAT configurations do not work.
Thanks, Steve T
On Sun, Jun 7, 2015 at 8:46 AM, Steve Totaro
Zitat von Steve Totaro:
Hi Steve,
My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but
direct in Internet. But maybe my Provider does a NAT, too…
Very strange is, that I have a very poorly audio-quality, if I use my
cellphone in my WLAN and connect to my Asterisk. With THE SAME USER, but from a PC in the same Network, the audio
quality is perfect.
Any idea?
Thanks Luca Bertoncello
(lucabert@lucabert.de)
Did you check which codecs are active? What does “sip show channelstats” say?
jg
Not without seeing some SIP debug output.
Thanks, Steve T
Zitat von Steve Totaro:
I’m currently not at home. If you say me which debug output you wish, I can send them as soon
I’ll be back…
Thanks Luca Bertoncello
(lucabert@lucabert.de)