[Asterisk-Users] Net2Phone / Vonage

[Russell Handorf]

Just got off the phone with Net2Phone; they now require 3 credentials to 
authenticate: account id, pin number, and MAC address. Any ideas?

Thanks

Russell Handorf wrote:

> I can cut and paste the log file from a reload right now, and provide 
> you with the other information when I get home after work:
>
> tmp*CLI> sip debug
> SIP Debugging Enabled
> tmp*CLI> reload
> Mar 21 14:52:42 NOTICE[23231]: indications.c:397 
> ast_unregister_indication_country: Removed default indication country 
> 'us'
> 11 headers, 0 lines
> Reliably Transmitting:
> REGISTER sip:sipvoiceline.net2phone.com SIP/2.0
> Via: SIP/2.0/UDP 38.115.19.11:5060;branch=z9hG4bK57947805
> From: 
> <sip:<snip-accountnumber>@sipvoiceline.net2phone.com>;tag=as7dc33a75
> To: <sip:<snip-accountnumber>@sipvoiceline.net2phone.com>
> Call-ID: 0306c6897a0807ea21158aef1749cbe0 at 127.0.0.1
> CSeq: 102 REGISTER
> User-Agent: INNOMEDIA_2PORT_ROUTER
> Expires: 120
> Contact: <sip:s at 38.115.19.11>
> Event: registration
> Content-Length: 0
>
> (no NAT) to 66.33.157.18:5060
> tmp*CLI>
>
> Sip read:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 38.115.19.11:5060;branch=z9hG4bK57947805
> From: 
> <sip:<snip-accountnumber>@sipvoiceline.net2phone.com>;tag=as7dc33a75
> To: <sip:<snip-accountnumber>@sipvoiceline.net2phone.com>
> Call-ID: 0306c6897a0807ea21158aef1749cbe0 at 127.0.0.1
> Expires: 3600
> CSeq: 102 REGISTER
> Contact: <sip:s at 38.115.19.11>
> Content-Length: 0
>
>
> 9 headers, 0 lines
> Mar 21 14:52:42 NOTICE[13089]: chan_sip.c:6797 handle_response: 
> Outbound Registration: Expiry for sipvoiceline.net2phone.com is 3600 
> sec (Scheduling reregistration in 3585000 ms)
> Destroying call '0306c6897a0807ea21158aef1749cbe0 at 127.0.0.1'
> 11 headers, 2 lines
> Reliably Transmitting:
> NOTIFY sip:bedroom2 at 192.168.5.143:5061 SIP/2.0
> Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK5be6f565;rport
> From: "asterisk" <sip:asterisk at 192.168.5.200>;tag=as5feb5ba4
> To: <sip:bedroom2 at 192.168.5.143:5061>
> Contact: <sip:asterisk at 192.168.5.200>
> Call-ID: 5b0c2dae5d7fd21b06b6e3c822cc4050 at 192.168.5.200
> CSeq: 102 NOTIFY
> User-Agent: INNOMEDIA_2PORT_ROUTER
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 42
>
> Messages-Waiting: no
> Voice-Message: 0/0
> (NAT) to 192.168.5.143:5061
> Scheduling destruction of call 
> '5b0c2dae5d7fd21b06b6e3c822cc4050 at 192.168.5.200' in 15000 ms
> tmp*CLI>
>
> Sip read:
> SIP/2.0 200 OK
> To: <sip:bedroom2 at 192.168.5.143:5061>
> From: "asterisk" <sip:asterisk at 192.168.5.200>;tag=as5feb5ba4
> Call-ID: 5b0c2dae5d7fd21b06b6e3c822cc4050 at 192.168.5.200
> CSeq: 102 NOTIFY
> Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK5be6f565;rport
> Server: Sipura/SPA2000-1.0.30
> Content-Length: 0
>
>
> 8 headers, 0 lines
> Destroying call '5b0c2dae5d7fd21b06b6e3c822cc4050 at 192.168.5.200'
> 11 headers, 2 lines
> Reliably Transmitting:
> NOTIFY sip:bedroom1 at 192.168.5.143:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK1fc74fe3;rport
> From: "asterisk" <sip:asterisk at 192.168.5.200>;tag=as01afe528
> To: <sip:bedroom1 at 192.168.5.143:5060>
> Contact: <sip:asterisk at 192.168.5.200>
> Call-ID: 046f554e56f323ce3cea8cdf2f2f3f9b at 192.168.5.200
> CSeq: 102 NOTIFY
> User-Agent: INNOMEDIA_2PORT_ROUTER
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 42
>
> Messages-Waiting: no
> Voice-Message: 0/0
> (NAT) to 192.168.5.143:5060
> Scheduling destruction of call 
> '046f554e56f323ce3cea8cdf2f2f3f9b at 192.168.5.200' in 15000 ms
> tmp*CLI>
>
> Sip read:
> SIP/2.0 200 OK
> To: <sip:bedroom1 at 192.168.5.143:5060>
> From: "asterisk" <sip:asterisk at 192.168.5.200>;tag=as01afe528
> Call-ID: 046f554e56f323ce3cea8cdf2f2f3f9b at 192.168.5.200
> CSeq: 102 NOTIFY
> Via: SIP/2.0/UDP 192.168.5.200:5060;branch=z9hG4bK1fc74fe3;rport
> Server: Sipura/SPA2000-1.0.30
> Content-Length: 0
>
>
> 8 headers, 0 lines
> Destroying call '046f554e56f323ce3cea8cdf2f2f3f9b at 192.168.5.200'
> Destroying call '490886b47b74afa62de6ee7324ef7c19 at 127.0.0.1'
> tmp*CLI> sip show peers
> Name/username    Host            Dyn Nat ACL Mask             Port     
> Status   net2phone/<snip>  66.33.157.18                
> 255.255.255.255  5060     Unmonitored
> rhandorf/rhando  (Unspecified)    D   N      255.255.255.255  0        
> Unmonitored
> bedroom2/bedroo  192.168.5.143    D   N      255.255.255.255  5061     
> Unmonitored
> bedroom1/bedroo  192.168.5.143    D   N      255.255.255.255  5060     
> Unmonitored
> 101/101          192.168.5.144    D          255.255.255.255  5060     
> Unmonitored
> 100/100          192.168.5.144    D          255.255.255.255  5060     
> Unmonitored
> tmp*CLI> sip no debug
> SIP Debugging Disabled
>
> when I call in, no packets even hit my server.
>
> Thanks,
> r
>
> kurt x wrote:
>
>> Russell,
>>
>> Do you have a SIP trace of the N2P call.  I would like to see if N2P
>> is getting the Invite and what it is replying with.
>>
>> Kurt
>>  
>>
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