[asterisk-users] can't call through voip provider

[Landy Landy]

> 
> Ok. I do NOT have ports 10000-20000 opened in. I guess I
> should try that and see if it works.
> 
> I will open ports 5060 - 5070 and 10000 - 100100 and do
> some test tonight. I will keep you posted.
> 

I ran this test and there was no difference.

I still can't get through. 

---
Retransmitting #5 (NAT) to 190.80.153.193:5060:
INVITE sip:18292574075 at optimumwireless.myvnc.com SIP/2.0
Via: SIP/2.0/UDP 190.80.153.193:5060;branch=z9hG4bK727987ef
Max-Forwards: 70
From: "102" <sip:77000 at 190.80.153.193>;tag=as23e02274
To: <sip:18292574000 at optimumwireless.myvnc.com>
Contact: <sip:77000 at 190.80.153.193>
Call-ID: 034bf0572cffb96f621211a8439aa9d7 at 190.80.153.193
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.5
Date: Thu, 19 Nov 2009 12:50:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 475

v=0
o=root 752676658 752676658 IN IP4 190.80.153.193
s=Asterisk PBX 1.6.1.5
c=IN IP4 190.80.153.193
t=0 0
m=audio 10026 RTP/AVP 0 3 8 112 5 10 7 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


I don't know why I don't see my provider's ip address. Isn't supposed to show in this debug?

Here's my sip.conf file again maybe you can catch an error or something I'm missing.

[voipprovider]
type=peer
host=208.78.163.3
username=77000
fromuser=77000
secret=77000
port=5060
dtmfmode=rfc2833
nat=route
insucure=port,invite
allow=all
careinvite=yes

Please helppppppp.