Hello All,
I’ve been with this problem for over two days straight, and run out of ideas to whats going on…
This is my scenario:
Asterisk Box --------> pfSense (Router&FW)--------------->INTERNET
192.168.1.5 ---->[In:192.168.1.1-Out:200.x.x.106]---->200.x.x.105
And I have a 1:1 NAT set up for my Asterisk box for management only (SSH,WEB):
192.168.1.5 <—> 200.x.x.108
I just bought an account at onesuite.com, they provided me with the typical configs for a SIP softphone plus my DID number, which I tried first with ZOIPER and works perfectly (incoming and outgoing calls)…!!
So now, all I did was set up this account on my existing Asterisk 1.6, and outgoing calls work perfectly, however, I cant get INCOMING calls to go through my Asterisk, and make an extension ring!!!
Here are the configs and some debugs:
--------------------------sip.conf----------------------------------
register => TWIITI2010-voip.onesuite.com:XxXxXxX@voip.onesuite.com/TWIITI2010-voip.onesuite.com
[onesuite]
dtmfmode=rfc2833
type=friend
authuser=TWIITI2010-voip.onesuite.com
username=TWIITI2010-voip.onesuite.com
secret=XxXxXxX
fromuser=TWIITI2010-voip.onesuite.com
fromdomain=voip.onesuite.com
host=voip.onesuite.com
callerid=Twiiti_USA
nat=yes
context=from-internal
call-limit=10
qualify=yes
insecure=port,invite
dynamic=yes
disallow=all
allow=ulaw
allow=alaw
---------------------sip_nat.conf----------------------------------
externip=200.x.x.108
localnet=192.168.1.0/255.255.255.0
localhost=192.168.1.5
nat=yes
---------------------extensions.conf----------------------------------
[from-internal]
include => sip-usa
include => sip-internacional
include => sip-incoming
[sip-usa]
exten => _9001NXXNXXXXXX,1,Dial(SIP/onesuite/${EXTEN:3},60,Tr)
exten => _9001NXXNXXXXXX,2,Playback(invalid)
exten => _9001NXXNXXXXXX,3,Hangup
exten => _9001NXXNXXXXXX,n+1,Hangup
[sip-internacional]
exten => _900[2-9].,1,Dial(SIP/onesuite/011${EXTEN:3},60,Tr)
exten => _900[2-9].,2,Playback(invalid)
exten => _900[2-9].,3,Hangup
exten => _900[2-9].,n+1,Hangup
[sip-incoming]
exten => TWIITI2010-voip.onesuite.com,1,Goto(SIP/300,s,1)
exten => TWIITI2010-voip.onesuite.com,2,Playback(invalid)
exten => TWIITI2010-voip.onesuite.com,3,Hangup
exten => TWIITI2010-voip.onesuite.com,n+1,Hangup
Some debugs:
localhost*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
voip.onesuite.com:5060 N TWIITI2010-v 23 Registered Tue, 13 Sep 2011 17:36:56
1 SIP registrations.
localhost*CLI>
<— SIP read from UDP:64.77.227.110:5060 —>
INVITE sip:TWIITI2010-voip.onesuite.com@200.x.x.108 SIP/2.0
Via: SIP/2.0/UDP 64.77.227.110:5060;branch=z9hG4bK1sansay238454430rdb5354
Via: SIP/2.0/UDP 10.1.100.90:5060;branch=z9hG4bK-d8754z-4f7d3850e032eb4e-1—d8754z-;rport=5060
Record-Route: sip:sansay238454430rdb5354@64.77.227.110:5060;lr;transport=udp
To: sip:TWIITI2010-voip.onesuite.com@200.x.x.108
From: sip:2134558960@64.77.227.110;tag=sansay238454430rdb5354
Call-ID: 68455473-0-625961700@64.77.227.114
CSeq: 1 INVITE
Contact: sip:2134558960@64.77.227.110:5060
Supported: timer,norefersub
Session-Expires: 600;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY, SUBSCRIBE, REGISTER
User-Agent: IVR TalkingSIP/3.10.1.0
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 228
v=0
o=Sansay-VSXi 188 1 IN IP4 10.1.100.87
s=Session Controller
c=IN IP4 64.77.227.110
t=0 0
m=audio 10714 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
— (17 headers 11 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 64.77.227.110 : 5060 (NAT)
Using INVITE request as basis request - 68455473-0-625961700@64.77.227.114
Found peer ‘OneSuite’ for ‘2134558960’ from 64.77.227.110:5060
<— Reliably Transmitting (NAT) to 64.77.227.110:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 64.77.227.110:5060;branch=z9hG4bK1sansay238454430rdb5354;received=64.77.227.110
Via: SIP/2.0/UDP 10.1.100.90:5060;branch=z9hG4bK-d8754z-4f7d3850e032eb4e-1—d8754z-;rport=5060
From: sip:2134558960@64.77.227.110;tag=sansay238454430rdb5354
To: sip:TWIITI2010-voip.onesuite.com@200.x.x.108;tag=as79257b41
Call-ID: 68455473-0-625961700@64.77.227.114
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="099f8574"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘68455473-0-625961700@64.77.227.114’ in 32000 ms (Method: INVITE)
localhost*CLI>
Retransmitting #1 (NAT) to 64.77.227.110:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 64.77.227.110:5060;branch=z9hG4bK1sansay238454430rdb5354;received=64.77.227.110
Via: SIP/2.0/UDP 10.1.100.90:5060;branch=z9hG4bK-d8754z-4f7d3850e032eb4e-1—d8754z-;rport=5060
From: sip:2134558960@64.77.227.110;tag=sansay238454430rdb5354
To: sip:TWIITI2010-voip.onesuite.com@200.x.x.108;tag=as79257b41
Call-ID: 68455473-0-625961700@64.77.227.114
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="099f8574"
Content-Length: 0
localhost*CLI>
<— SIP read from UDP:64.77.227.110:5060 —>
ACK sip:TWIITI2010-voip.onesuite.com@200.x.x.108 SIP/2.0
Via: SIP/2.0/UDP 64.77.227.110:5060;branch=z9hG4bK1sansay238454430rdb5354
To: sip:TWIITI2010-voip.onesuite.com@200.x.x.108;tag=as79257b41
From: sip:2134558960@64.77.227.110;tag=sansay238454430rdb5354
Call-ID: 68455473-0-625961700@64.77.227.114
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
Everytime I try to dial my DID number, I get these errors!..
PLEASE HELP!