Nothing. I don't know what in the world is going on with my setup. Here's my FORWARD rules: eth0 = external nic, eth1 = lan
0 0 ACCEPT udp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 udp dpts:5060:5070 0 0 ACCEPT udp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 udp dpts:10000:10100 1 62 ACCEPT udp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 udp dpts:5060:5070 36 2372 ACCEPT udp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 udp dpts:10000:10100 0 0 ACCEPT tcp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 tcp dpts:5060:5070 0 0 ACCEPT tcp -- eth0 eth1 0.0.0.0/0 0.0.0.0/0 tcp dpts:10000:10100 0 0 ACCEPT tcp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 tcp dpts:5060:5070 3 144 ACCEPT tcp -- eth1 eth0 0.0.0.0/0 0.0.0.0/0 tcp dpts:10000:10100 and now the debug: etransmitting #5 (NAT) to 190.80.152.200:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.152.200:5060;branch=z9hG4bK794de7aa;rport Max-Forwards: 70 From: "102" <sip:77...@190.80.152.200>;tag=as5084570c To: <sip:18292574...@optimumwireless.myvnc.com> Contact: <sip:77...@190.80.152.200> Call-ID: 22569d3b767276276c6c65c84b314...@190.80.152.200 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 22:53:06 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 135722140 135722140 IN IP4 190.80.152.200 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.200 t=0 0 m=audio 10074 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'm already frustrated with this. --- On Thu, 11/19/09, Warren Selby <wcse...@selbytech.com> wrote: > From: Warren Selby <wcse...@selbytech.com> > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Date: Thursday, November 19, 2009, 5:11 PM > On Thu, Nov 19, > 2009 at 3:36 PM, Landy Landy <landysacco...@yahoo.com> > wrote: > > Can someone please share with me a sip configuration to > connect an asterisk server to a voip provider since my > configuration isn't working for me. > > > > thanks. > > > > > Who is your voipprovider? Did they give you the settings > you're using in your sip.conf? Also, you've got > some typos in your sip config (insucure = insecure, > careinvite = canreinvite). You could try something like > this: > > > [voipprovider] > > type=peer > > host=208.78.163.3 > > username=77000 > > fromuser=77000 > > secret=77000 > > port=5060 > > dtmfmode=rfc2833 > > nat=yes > canreinvite=yes > > insecure=very > disallow=all > allow=ulaw > allow=alaw > > > > > > -- > Thanks, > --Warren Selby > http://www.selbytech.com > > > -----Inline Attachment Follows----- > > _______________________________________________ > -- 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 _______________________________________________ -- 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