[asterisk-users] Re: Vonage SIP access via asterisk?

Thu Dec 14 14:23:24 MST 2006

I can dial out via FWD because the login is in the session.

Vonage requires a register even for outbound.
My understanding is that they log the register and then any call from that IP is from that user.
This is why I can't dial out vonage.

The root cause is that sip is not registering at all.

I assume the firewall is OK, because I can make outbound SIP calls to FWD.

Is there a way to force a registry so that I can capture the packets?

-- 
-- 
Steven

http://www.glimasoutheast.org

"Steven" <asterisk at tescogroup.com> wrote in message news:elsf3o$qpq$1 at sea.gmane.org...
> 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" <asterisk at tescogroup.com> wrote in message news:elc6ud$13h$1 at sea.gmane.org...
>> 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:124864xxxxx at sphone.vopr.vonage.net:5061 SIP/2.0
>> Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport
>> From: "SteveB TEST" <sip:1xvonagenum at sphone.vopr.vonage.net>;tag=as35e23a92
>> To: <sip:124864xxxxx at sphone.vopr.vonage.net:5061>
>> Contact: <sip:1xvonagenum at 64.118.155.160>
>> Call-ID: 6af934582eeb68b5439e917d2f47ca4c at sphone.vopr.vonage.net
>> 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 124864xxxxx at 1xvonagenum
>> 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:1xvonagenum at sphone.vopr.vonage.net>;tag=as35e23a92
>> To: <sip:124864xxxxx at sphone.vopr.vonage.net:5061>
>> Call-ID: 6af934582eeb68b5439e917d2f47ca4c at sphone.vopr.vonage.net
>> CSeq: 102 INVITE
>> Max-Forwards: 15
>> Content-Length: 0
>>
>>
>> --- (8 headers 0 lines) ---
>> Transmitting (NAT) to 216.115.20.41:5061:
>> ACK sip:124864xxxxx at sphone.vopr.vonage.net:5061 SIP/2.0
>> Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport
>> From: "SteveB TEST" <sip:1xvonagenum at sphone.vopr.vonage.net>;tag=as35e23a92
>> To: <sip:124864xxxxx at sphone.vopr.vonage.net:5061>
>> Contact: <sip:1xvonagenum at 64.118.155.160>
>> Call-ID: 6af934582eeb68b5439e917d2f47ca4c at sphone.vopr.vonage.net
>> CSeq: 102 ACK
>> User-Agent: Asterisk PBX
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> -- 
>> -- 
>> Steven
>>
>> http://www.glimasoutheast.org
>>
>>
>>
>> "Al Bochter" <Al.Bochter at bochterservices.com> wrote in message news:45799470.1000400 at bochterservices.com...
>>> 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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>>>>
>>>>
>>>>
>>>>----------------------------------------------------
>>>>Inbound (clean). Database: 0654-1, 12/07/2006 - 12/8/2006 11:10:08 AM
>>>>
>>>>
>>>>
>>>>
>>>>
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>>
>>
>>
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