This may not be vonage related as it appears that I can not register with any sip servers. I tried FWD and also get a black "sip show registry"
Could it be a firewall issue? I am running IP tables on the computer which is on the internet with no NAT. Asterisk 1.2.13 I have allow outbound all. Allow inbound 5060, IAX and RTP. -- -- Steven http://www.glimasoutheast.org "Steven" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > That and any other ref.s I have found give me a 404 error when dialing out. > > My Sip show registry is also empty. > > ref: > We're at 64.x.x.x port 12146 > Adding codec 0x4 (ulaw) to SDP > Adding codec 0x8 (alaw) to SDP > Adding codec 0x1 (g723) to SDP > Adding codec 0x2 (gsm) to SDP > Adding codec 0x10 (g726) to SDP > Adding codec 0x20 (adpcm) to SDP > Adding codec 0x40 (slin) to SDP > Adding codec 0x80 (lpc10) to SDP > Adding codec 0x100 (g729) to SDP > Adding codec 0x200 (speex) to SDP > Adding codec 0x400 (ilbc) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > 13 headers, 21 lines > Reliably Transmitting (NAT) to 216.115.20.41:5061: > INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0 > Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport > From: "SteveB TEST" <sip:[EMAIL PROTECTED]>;tag=as35e23a92 > To: <sip:[EMAIL PROTECTED]:5061> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Fri, 08 Dec 2006 17:15:22 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 494 > > v=0 > o=root 9983 9983 IN IP4 64.118.155.160 > s=session > c=IN IP4 64.118.155.160 > t=0 0 > m=audio 12146 RTP/AVP 0 8 4 3 111 5 10 7 18 110 97 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:111 G726-32/8000 > a=rtpmap:5 DVI4/8000 > a=rtpmap:10 L16/8000 > a=rtpmap:7 LPC/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:110 speex/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > -- Called [EMAIL PROTECTED] > tg05*CLI> > <-- SIP read from 216.115.20.41:5061: > SIP/2.0 404 Not Found > Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport > From: "SteveB TEST" <sip:[EMAIL PROTECTED]>;tag=as35e23a92 > To: <sip:[EMAIL PROTECTED]:5061> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > Max-Forwards: 15 > Content-Length: 0 > > > --- (8 headers 0 lines) --- > Transmitting (NAT) to 216.115.20.41:5061: > ACK sip:[EMAIL PROTECTED]:5061 SIP/2.0 > Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport > From: "SteveB TEST" <sip:[EMAIL PROTECTED]>;tag=as35e23a92 > To: <sip:[EMAIL PROTECTED]:5061> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > -- > -- > Steven > > http://www.glimasoutheast.org > > > > "Al Bochter" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] >> http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+6#621VonageBusinessPlusandVonageSoftphoneb >> >> Best regards, >> >> Al Bochter >> Bochter Services >> http://www.BochterServices.com/?t=Email >> >> (VOIP PBX) 1-866-638-1254 >> >> (Voip PBX) Free World DialUp: 780-217 >> WebSite: http://www.freeworlddialup.com/ >> >> We have Toll Free DID's instock >> * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * >> http://www.bochterservices.com/?t=TF(NM)did >> >> BUY Coins, Silver and Gold >> http://www.bochterservices.com/?j=gold&t=email >> >> For new and used security items >> http://www.bochterservices.com/?j=store&t=email_security >> >> >> >> BerkHolz, Steven wrote: >> >>>Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) >>> >>>I just signed up to test their service and they sent me a Number, Proxy, >>>port and password. >>> >>>Every reference I have tried leaves me with a 404 error coming from Vonage. >>> >>>If you have a working setup, please post some config references. >>> >>> >>> Thank You, >>>Steven BerkHolz >>> >>> >>> >>>Soon to be known as HIROTEC AMERICA >>>www.hirotecamerica.com >>>_______________________________________________ >>>--Bandwidth and Colocation provided by Easynews.com -- >>> >>>asterisk-users mailing list >>>To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >>> >>>---------------------------------------------------- >>>Inbound (clean). Database: 0654-1, 12/07/2006 - 12/8/2006 11:10:08 AM >>> >>> >>> >>> >>> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
