(((Trunk/INfo)))) All other areas are left blank!
Outbound Caller ID
xxx xxx xxxx
(((Trunk Name)))
vtwhite
(((Peer Details)))
allow=ulaw
canreinvite=yes
context=from-pstn
disallow=all
dtmfmode=rfc2833
fromdomain=sip.vtwhite.com
host=sip.vtwhite.com
insecure=invite,port
nat=yes
qualify=yes&yes
type=friend
(((Dial Rules)))
352|NXXXXXX
**************************************************************
(((Under Extensions))) Everything else is left blank or default
(((Display Name)))
test
(((Device Options)))
secret 123456
dtmfmode rfc2833
canreinvite no
context from-internal
host dynamic
type friend
nat yes
port 5060
qualify yes
callgroup
pickupgroup
disallow
allow
dial sip/100
accountcode
mailbox 100@device
******************************************************************
(((Under OUtbound Routes)))) Everything else is left blank or default
(((Route Name)))
outbound
(((Dial Patterns)))
011.
1800NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX
352|NXXXXXX
NXXNXXXXXX
NXXXXXX
((((Trunk Sequence)))
sip/vtwhite
********************************************************************
((((sip_nat.conf))))
((((This File I edited it manually as per suggestion from one of the members in here)))
nat=yes
localnet=192.168.1.0/255.255.255.0
externip=76.4.108.63
externrefresh=10
*********************************************************************
(((I have setup debug screen and below is what I get when its just idle, this scrolls every 30 seconds or so)))
asterisk1*CLI>
<-- SIP read from 216.120.251.172:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK73e728be;rport=5060
From: "Unknown" <sip:Unknown@xxx.xxx.xxx.xxx>;tag=as763380b5
To: <sip:sip.vtwhite.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.1119
Call-ID: 12b002ac2f26e4ac5fdf9c0f1f5d2ff1@xxx.xxx.xxx.xxx
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Support:
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 216.120.251.172:5060 "Noisy feedback tells: pid=8511 req_src_ip=xxx.xxx.xxx.xxx req_src_port=5060 in_uri=sip:sip.vtwhite.com out_uri=sip:sip.vtwhite.com via_cnt==1"
**************************************************************************
(((((Below is what actually happens when I place an outbound call, some phone numbers seem to dial out but noone picks up, and I know someone should be there, other numbers immediately give me circuits are buisy try again later. There is also two IP numbers that do not seem to belong in this debug (64.118.82.166 and 4.68.251.11)As an amarture, I am not sure if thats normal or not.
--- (9 headers 0 lines) ---
asterisk1*CLI>
<-- SIP read from 216.120.251.172:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3872e247;rport=5060
Record-Route: <sip:64.118.82.166;lr=on;ftag=as3bb8ed4a>,<sip:216.120.251.172;lr=on;ftag=as3bb8ed4a>
From: "My VOIP PHone Number" <sip:My VOIP PHone Number@sip.vtwhite.com>;tag=as3bb8ed4a
To: <sip:7 DIGIT NUMBER i DIALED@sip.vtwhite.com>;tag=VPST506071629579
Call-ID: 2a9e78907ca9682b79d60946544e347e@sip.vtwhite.com
CSeq: 102 INVITE
Contact: <sip:+7 DIGIT NUMBER i DIALED@4.68.251.11:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 182
v=0
o=- 1180402670 1180402671 IN IP4 209.247.5.190
s=-
c=IN IP4 209.247.5.190
t=0 0
m=audio 64464 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.247.5.190:64464
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
12 headers, 0 lines
Reliably Transmitting (NAT) to 216.120.251.172:5060:
OPTIONS sip:sip.vtwhite.com SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK61905576;rport
From: "Unknown" <sip:Unknown@xxx.xxx.xxx.xxx>;tag=as6a1311e3
To: <sip:sip.vtwhite.com>
Contact: <sip:Unknown@xxx.xxx.xxx.xxx>
Call-ID: 6e61d8cd3e7364ce456d4c0b6de60f23@xxx.xxx.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 29 May 2007 01:38:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0