[Asterisk-Users] SIP behind NAT (*sigh*)

[Martin Pycko]

Well your debug only shows that the phone didn't register itself with
asterisk.

asterisk transmits:
"SIP/2.0 407 Proxy Authentication Required"

Martin
On Thu, 15 May 2003, Florian Overkamp wrote:

> Hi guys,
>
> sorry to be iterating this on the list once more, but I'm not able to get
> this stuff to work as I'd expect. So far, I've always managed to keep it
> out of NAT environments :->
>
> My home LAN is NATed by a simple Draytek router.
>
> In the home LAN is an ATA186 with SIP. On the internet (public) is an
> Asterisk server.
>
> I have nat=yes in the sip.conf and the connectmode is set to look for the
> Via header.
>
> Registration works like a charm, and if I dial in from the PSTN to the ATA
> the phone rings properly. However, it doesn't seem to be able to start an
> RTP stream or something, because once I try to dial, it gives me a
> busy/congestion tone after a couple of tries (looking at the debug info).
> The context is set properly, and I have tried to enable port-forwarding of
> the RTP port toward the ATA, but no luck so far..
>
> Maybe one of you has an idea ?
>
>
>
>
> vectra*CLI>
> Sip read: >
> INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To:  <sip:0534280105 at 217.114.97.249;user=phone>
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> Contact:  <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Expires: 300
> Content-Length: 253
> Content-Type: application/sdp
>
> v=0
> o=ata1-1 33968 33968 IN IP4 130.89.224.240
> s=ATA186 Call
> c=IN IP4 130.89.224.0
> t=0 0
> m=audio 16384 RTP/AVP 0 4 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:4 G723/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 11 headers, 11 lines
> Using latest request as basis request
> Sending to 130.89.224.240 : 5060 (non-NAT)
> Capabilities: us - 14, them - 13, combined - 12
> Non-codec capabilities: us - 1, them - 1, combined - 1
> Reliably Transmitting (NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Contact:
> Proxy-Authenticate: Digest realm="asterisk", nonce="1e537e10"
> Content-Length: 0
>
>
>   to 130.89.224.240:5060
> Sip read: >
> ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 ACK
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Content-Length: 0
>
>
> 8 headers, 0 lines
> Sip read: >
> INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To:  <sip:0534280105 at 217.114.97.249;user=phone>
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> Contact:  <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Expires: 300
> Content-Length: 253
> Content-Type: application/sdp
>
> v=0
> o=ata1-1 33968 33968 IN IP4 130.89.224.240
> s=ATA186 Call
> c=IN IP4 130.89.224.0
> t=0 0
> m=audio 16384 RTP/AVP 0 4 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:4 G723/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 11 headers, 11 lines
> Ignoring this request
> Reliably Transmitting (NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Contact:
> Proxy-Authenticate: Digest realm="asterisk", nonce="26d61be9"
> Content-Length: 0
>
>
>   to 130.89.224.240:5060
> Sip read: >
> ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 ACK
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Content-Length: 0
>
>
> 8 headers, 0 lines
> Sip read: >
> INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To:  <sip:0534280105 at 217.114.97.249;user=phone>
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> Contact:  <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Expires: 300
> Content-Length: 253
> Content-Type: application/sdp
>
> v=0
> o=ata1-1 33968 33968 IN IP4 130.89.224.240
> s=ATA186 Call
> c=IN IP4 130.89.224.0
> t=0 0
> m=audio 16384 RTP/AVP 0 4 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:4 G723/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 11 headers, 11 lines
> Ignoring this request
> Reliably Transmitting (NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Contact:
> Proxy-Authenticate: Digest realm="asterisk", nonce="5a6ef345"
> Content-Length: 0
>
>
>   to 130.89.224.240:5060
> Sip read: >
> ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 ACK
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Content-Length: 0
>
>
> 8 headers, 0 lines
> Sip read: >
> INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To:  <sip:0534280105 at 217.114.97.249;user=phone>
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> Contact:  <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Expires: 300
> Content-Length: 253
> Content-Type: application/sdp
>
> v=0
> o=ata1-1 33968 33968 IN IP4 130.89.224.240
> s=ATA186 Call
> c=IN IP4 130.89.224.0
> t=0 0
> m=audio 16384 RTP/AVP 0 4 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:4 G723/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 11 headers, 11 lines
> Ignoring this request
> Reliably Transmitting (NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Contact:
> Proxy-Authenticate: Digest realm="asterisk", nonce="02af8ccd"
> Content-Length: 0
>
>
>   to 130.89.224.240:5060
> Sip read: >
> ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 ACK
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Content-Length: 0
>
>
> 8 headers, 0 lines
> Sip read: >
> INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To:  <sip:0534280105 at 217.114.97.249;user=phone>
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> Contact:  <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Expires: 300
> Content-Length: 253
> Content-Type: application/sdp
>
> v=0
> o=ata1-1 33968 33968 IN IP4 130.89.224.240
> s=ATA186 Call
> c=IN IP4 130.89.224.0
> t=0 0
> m=audio 16384 RTP/AVP 0 4 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:4 G723/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 11 headers, 11 lines
> Ignoring this request
> Reliably Transmitting (NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Contact:
> Proxy-Authenticate: Digest realm="asterisk", nonce="02cbd912"
> Content-Length: 0
>
>
>   to 130.89.224.240:5060
> Sip read: >
> ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 ACK
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Content-Length: 0
>
>
> 8 headers, 0 lines
> Sip read: >
> INVITE sip:0534280105 at 217.114.97.249;user=phone SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To:  <sip:0534280105 at 217.114.97.249;user=phone>
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> Contact:  <sip:ata1-1 at 130.89.224.0:5060;transport=udp>
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Expires: 300
> Content-Length: 253
> Content-Type: application/sdp
>
> v=0
> o=ata1-1 33968 33968 IN IP4 130.89.224.240
> s=ATA186 Call
> c=IN IP4 130.89.224.0
> t=0 0
> m=audio 16384 RTP/AVP 0 4 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:4 G723/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 11 headers, 11 lines
> Ignoring this request
> Reliably Transmitting (NAT):
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Contact:
> Proxy-Authenticate: Digest realm="asterisk", nonce="1be02d93"
> Content-Length: 0
>
>
>   to 130.89.224.240:5060
> Sip read: >
> ACK sip:0534280105 at 217.114.97.249:5060 SIP/2.0
> Via: SIP/2.0/UDP 130.89.224.240:5060;received=130.89.224.240
> From: sip:ata1-1 at 217.114.97.249;tag=2733832243
> To: <sip:0534280105 at 217.114.97.249;user=phone>;tag=as7503b8bb
> Call-ID: 855024110 at 130.89.224.240
> CSeq: 1 ACK
> User-Agent: Cisco ATA  v2.15 ata18x (020927a)
> Content-Length: 0
>
>
> 8 headers, 0 lines
>
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