Sorry to bother you again with my problem but, is that I can't figure out what's going on with my setup. I have no idea of why my asterisk server is not communicating with my provider's. I've searched, googled, and can't find my solution. I've followed many tutorials but can't get anywhere.
--- On Thu, 11/19/09, Landy Landy <landysacco...@yahoo.com> wrote: > From: Landy Landy <landysacco...@yahoo.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, 5:53 PM > 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 > _______________________________________________ -- 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