I’m new to Asterisk and hope someone can help me to solve a trouble I’ve been spending days on 
My network configuration is depicted herehttp://it.tinypic.com/view.php?pic=dfk3up&s=8. Basically a cable modem which feeds a nat router/firewall ending up in 2 vlan.
On the lan side I have Asterisk 11.8.1 running on a small equipment (OpenWrt) and some sip clients (ip phones and softphones). Calls between phones on the lan side are ok.
Asterisk has a trunk (peer) towards a voip provider and registers ok. Calls from pstn to voip provider’s number are ok: the extensions ring and audio is ok in both directions.
Troubles arise when I try to place a call from one of the phones to pstn (or to a mobile phone) using the provider’s trunk. Pstn phone rings but I have no ringback tone and, if answer the call, no audio both ways.
The cable modem has the firewall as its DMZ host, forwarding all traffic incoming on ex.te.rn.ip to the firewall and the firewall forwards utp ports 10000-20000 to Asterisk’s host.
It doesn’t forward udp 5060 port because there is nothing outside that needs to register to Asterisk, though I tried forwarding that port too but nothing changed.
Setting in a softphone on the same lan the voip provider’s parameters gets everything ok: a call to pstn gets ringback tone and audio is ok, so I can exclude any provider’s issue.
This is the relevant part of sip.conf
[general]
language = it
tonezone = it
context=public
allowguest=no
srvlookup=yes
transport=udp
keepalive=30
udpbindaddr=0.0.0.0
tcpenable=no
externaddr = ex.te.rn.ip
localnet=192.168.30.0/24
dtmfmode=auto
disallow = all
allow = alaw
allow = ulaw
allow = g726
rtpkeepalive = 30
nat = force_rport, comedia
register => *********:***********@voip.eutelia.it/eutelia_ext
[eutelia]
type = peer
context = eutelia
defaultuser = ************
fromuser = ************
secret = *************
host = voip.eutelia.it
port = 5060
qualify = yes
insecure = invite,port
canreinvite = no
directmedia = no
[utenti](!)
type=friend
context=LocalSets
host=dynamic
qualify=yes
secret=*******
[41](utenti)
fullname=Papero
[42](utenti)
fullname=Gatto
[43](utenti)
fullname=Topo
[44](utenti)
fullname=Conigliortp.conf
[general]
rtpstart=14940
rtpend=14990
stunaddr=stun.voip.eutelia.it:3478sip show settings
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm OpenWrt
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 11.8.1
SDP Session Name: Asterisk PBX 11.8.1
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Enabled using externaddr
Externhost: <none>
Externaddr: ex.te.rn.ip:0
Externrefresh: 10
Localnet: 192.168.30.0/255.255.255.0
Global Signalling Settings:
---------------------------
Codecs: (ulaw|alaw|g726)
Codec Order: alaw:20,ulaw:20,g726:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 30
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: public
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: auto
Qualify: 0
Keepalive: 30
Use ClientCode: No
Progress inband: Never
Language: it
Tone zone: it
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisklog.txt: https://www.dropbox.com/s/0ignum8di89nnug/log.txt
Any help appreciated. Thank you.
) in the same network environment and… it works flawlessly;
