[Asterisk-Users] New Firefly version

[Adam Hart]

the log looks legit except why does asterisk have a different IP in the 
contact compared to the 'to' address.

I can connect successfully to my asterisk server and FWD - can anyone 
give me sip access to a asterisk server that firefly doesn't work on?

miguel at amplanet.com.br wrote:
> Why all the time the firefly show me the message: Sip registration failed
> for the network Home (407). 
> 
> The server, username and password are correct. I'm using the default RTP
> port 5000 in the SIP tab.
> 
> Using the SJPhone I can register; using the firefly I can call any
> registered number, but I can't register. 
> 
> On asterisk console:
> 
> Sip read:
> REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0
> To: <sip:2003 at 192.168.199.3:5060;transport=udp>
> From: <sip:2003 at 192.168.199.121:5060>;tag=5a1c4f36
> Via: SIP/2.0/UDP
> 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport
> Call-ID: c90fa011e82acf3e
> CSeq: 1 REGISTER
> Contact: <sip:2003 at 192.168.199.121:5060>
> Expires: 3600
> Max-Forwards: 70
> User-Agent: Firefly
> Content-Length: 0
> 
> 
> 11 headers, 0 lines
> Using latest request as basis request
> Sending to 192.168.199.121 : 5060 (non-NAT)
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport
> From: <sip:2003 at 192.168.199.121:5060>;tag=5a1c4f36
> To: <sip:2003 at 192.168.199.3:5060;transport=udp>;tag=as7843e908
> Call-ID: c90fa011e82acf3e
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2003 at 192.168.199.4>
> Content-Length: 0
> 
> 
>  to 192.168.199.121:5060
> Transmitting (no NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
> 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport
> From: <sip:2003 at 192.168.199.121:5060>;tag=5a1c4f36
> To: <sip:2003 at 192.168.199.3:5060;transport=udp>;tag=as7843e908
> Call-ID: c90fa011e82acf3e
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2003 at 192.168.199.4>
> Proxy-Authenticate: Digest realm="asterisk", nonce="38165263"
> Content-Length: 0
> 
> 
>  to 192.168.199.121:5060
> SAMPLANET1*CLI>
> 
> Sip read:
> REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0
> To: <sip:2003 at 192.168.199.3:5060;transport=udp>
> From: <sip:2003 at 192.168.199.121:5060>;tag=6c3de14a
> Via: SIP/2.0/UDP
> 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport
> Call-ID: c90fa011e82acf3e
> CSeq: 1 REGISTER
> Contact: <sip:2003 at 192.168.199.121:5060>
> Expires: 3600
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username=2003,realm="asterisk",nonce="38165263",uri="sip:192.168.199.3:5060;
> transport=udp",response="ec0afc0a2b13a725aa40b5c311c396d8",algorithm=MD5
> User-Agent: Firefly
> Content-Length: 0
> 
> 
> 12 headers, 0 lines
> Using latest request as basis request
> Sending to 192.168.199.121 : 5060 (non-NAT)
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport
> From: <sip:2003 at 192.168.199.121:5060>;tag=6c3de14a
> To: <sip:2003 at 192.168.199.3:5060;transport=udp>;tag=as7843e908
> Call-ID: c90fa011e82acf3e
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2003 at 192.168.199.4>
> Content-Length: 0
> 
> 
>  to 192.168.199.121:5060
> Transmitting (no NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
> 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport
> From: <sip:2003 at 192.168.199.121:5060>;tag=6c3de14a
> To: <sip:2003 at 192.168.199.3:5060;transport=udp>;tag=as7843e908
> Call-ID: c90fa011e82acf3e
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2003 at 192.168.199.4>
> Proxy-Authenticate: Digest realm="asterisk", nonce="38165263"
> Content-Length: 0
> 
> 
>  to 192.168.199.121:5060 
> 
> 
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