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. --- 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, 7:51 AM > > > > > 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 > _______________________________________________ -- 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