Hello. I have my server running for about 30 days. Every time I did some changes to my sip.conf file I did reload in the cli. I thought this would change the new values. Somehow it wasn't. I decided to do a restart now and that used my new settings. The same settings I've been posting here the past week and weren't working. After restarting asterisk I'm able to use my provider via asterisk to make calls.
I would like to thank those who helped me. --- On Fri, 11/20/09, 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: Friday, November 20, 2009, 8:53 AM > 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 > _______________________________________________ -- 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
