> > 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:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.153.193:5060;branch=z9hG4bK727987ef Max-Forwards: 70 From: "102" <sip:77...@190.80.153.193>;tag=as23e02274 To: <sip:18292574...@optimumwireless.myvnc.com> Contact: <sip:77...@190.80.153.193> Call-ID: 034bf0572cffb96f621211a8439aa...@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. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users