(((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" ;tag=as763380b5 To: ;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: , From: "My VOIP PHone Number" ;tag=as3bb8ed4a To: ;tag=VPST506071629579 Call-ID: 2a9e78907ca9682b79d60946544e347e@sip.vtwhite.com CSeq: 102 INVITE Contact: 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" ;tag=as6a1311e3 To: Contact: 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