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
>>_______________________________________________
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>>
>>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
>>
>>
>>
>>
>>
> _______________________________________________
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