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@217.114.97.249;user=phone SIP/2.0 Via: SIP/2.0/UDP 130.89.224.240:5060 From: sip:ata1-1@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone> Call-ID: 855024110@130.89.224.240 CSeq: 1 INVITE Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@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@217.114.97.249;user=phone SIP/2.0 Via: SIP/2.0/UDP 130.89.224.240:5060 From: sip:ata1-1@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone> Call-ID: 855024110@130.89.224.240 CSeq: 1 INVITE Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@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@217.114.97.249;user=phone SIP/2.0 Via: SIP/2.0/UDP 130.89.224.240:5060 From: sip:ata1-1@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone> Call-ID: 855024110@130.89.224.240 CSeq: 1 INVITE Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@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@217.114.97.249;user=phone SIP/2.0 Via: SIP/2.0/UDP 130.89.224.240:5060 From: sip:ata1-1@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone> Call-ID: 855024110@130.89.224.240 CSeq: 1 INVITE Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@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@217.114.97.249;user=phone SIP/2.0 Via: SIP/2.0/UDP 130.89.224.240:5060 From: sip:ata1-1@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone> Call-ID: 855024110@130.89.224.240 CSeq: 1 INVITE Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@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@217.114.97.249;user=phone SIP/2.0 Via: SIP/2.0/UDP 130.89.224.240:5060 From: sip:ata1-1@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone> Call-ID: 855024110@130.89.224.240 CSeq: 1 INVITE Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb Call-ID: 855024110@130.89.224.240 CSeq: 1 ACK User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 8 headers, 0 lines
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@217.114.97.249;user=phone SIP/2.0 > Via: SIP/2.0/UDP 130.89.224.240:5060 > From: sip:ata1-1@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone> > Call-ID: 855024110@130.89.224.240 > CSeq: 1 INVITE > Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@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@217.114.97.249;user=phone SIP/2.0 > Via: SIP/2.0/UDP 130.89.224.240:5060 > From: sip:ata1-1@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone> > Call-ID: 855024110@130.89.224.240 > CSeq: 1 INVITE > Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@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@217.114.97.249;user=phone SIP/2.0 > Via: SIP/2.0/UDP 130.89.224.240:5060 > From: sip:ata1-1@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone> > Call-ID: 855024110@130.89.224.240 > CSeq: 1 INVITE > Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@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@217.114.97.249;user=phone SIP/2.0 > Via: SIP/2.0/UDP 130.89.224.240:5060 > From: sip:ata1-1@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone> > Call-ID: 855024110@130.89.224.240 > CSeq: 1 INVITE > Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@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@217.114.97.249;user=phone SIP/2.0 > Via: SIP/2.0/UDP 130.89.224.240:5060 > From: sip:ata1-1@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone> > Call-ID: 855024110@130.89.224.240 > CSeq: 1 INVITE > Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@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@217.114.97.249;user=phone SIP/2.0 > Via: SIP/2.0/UDP 130.89.224.240:5060 > From: sip:ata1-1@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone> > Call-ID: 855024110@130.89.224.240 > CSeq: 1 INVITE > Contact: <sip:ata1-1@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@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@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@217.114.97.249;tag=2733832243 > To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb > Call-ID: 855024110@130.89.224.240 > CSeq: 1 ACK > User-Agent: Cisco ATA v2.15 ata18x (020927a) > Content-Length: 0 > > > 8 headers, 0 lines > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Cristian Rodriguez
2003-May-15 14:29 UTC
[Asterisk-Users] fxs no dial tone on my USB device
Hello, Yes I checked the mail list archives Yes I have the latest CVS code I cannot get a dial tone from my USB FXS device. In my search through the list achives I found that it has been an issue with some others. I ensured the modules where loaded in the right order. I managed to get sound when I press the keys on the phone, but still no dialtone. I used the astinstall script that came with my dev lite kit. (The FXO PCI board is working fine) I tried unloading the modules and reloading them then ztcfg but still no dialtone. I am trying the install on a new box to elimitate the USB controller as one of the posibilities. I am using the sample configs. Perhaps I have not setup my extensions properly. PII 350 64 meg ram redhat 9.0 dev lite kit. USB usb-ohci.c: usb-00:09.0, OPTi Inc. 82C861 Can someone please help.
Brancaleoni Matteo
2003-May-15 15:03 UTC
[Asterisk-Users] fxs no dial tone on my USB device
asterisk starts ok? matteo Il gio, 2003-05-15 alle 23:29, Cristian Rodriguez ha scritto:> Hello, > > Yes I checked the mail list archives > Yes I have the latest CVS code > > I cannot get a dial tone from my USB FXS device. > > In my search through the list achives I found that it has been an issue with > some others. > > I ensured the modules where loaded in the right order. I managed to get > sound when I press the keys on the phone, but still no dialtone. > I used the astinstall script that came with my dev lite kit. (The FXO PCI > board is working fine) > > I tried unloading the modules and reloading them then ztcfg but still no > dialtone. > > I am trying the install on a new box to elimitate the USB controller as one > of the posibilities. > > I am using the sample configs. Perhaps I have not setup my extensions > properly. > > PII 350 64 meg ram redhat 9.0 dev lite kit. USB usb-ohci.c: usb-00:09.0, > OPTi Inc. 82C861 > > Can someone please help. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
When you type "zap show channels" on the asterisk console do you see your USB channel ? Martin On 16 May 2003, Brancaleoni Matteo wrote:> asterisk starts ok? > > matteo > > Il gio, 2003-05-15 alle 23:29, Cristian Rodriguez ha scritto: > > Hello, > > > > Yes I checked the mail list archives > > Yes I have the latest CVS code > > > > I cannot get a dial tone from my USB FXS device. > > > > In my search through the list achives I found that it has been an issue with > > some others. > > > > I ensured the modules where loaded in the right order. I managed to get > > sound when I press the keys on the phone, but still no dialtone. > > I used the astinstall script that came with my dev lite kit. (The FXO PCI > > board is working fine) > > > > I tried unloading the modules and reloading them then ztcfg but still no > > dialtone. > > > > I am trying the install on a new box to elimitate the USB controller as one > > of the posibilities. > > > > I am using the sample configs. Perhaps I have not setup my extensions > > properly. > > > > PII 350 64 meg ram redhat 9.0 dev lite kit. USB usb-ohci.c: usb-00:09.0, > > OPTi Inc. 82C861 > > > > Can someone please help. > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Cristian Rodriguez
2003-May-15 19:33 UTC
[Asterisk-Users] fxs no dial tone on my USB device
No I do not see my USB channel. I only see my inbound. ----- Original Message ----- From: "Martin Pycko" <martinp@digium.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, May 15, 2003 3:05 PM Subject: Re: [Asterisk-Users] fxs no dial tone on my USB device> When you type "zap show channels" on the asterisk console do you see your > USB channel ? > > Martin > > On 16 May 2003, Brancaleoni Matteo wrote: > > > asterisk starts ok? > > > > matteo > > > > Il gio, 2003-05-15 alle 23:29, Cristian Rodriguez ha scritto: > > > Hello, > > > > > > Yes I checked the mail list archives > > > Yes I have the latest CVS code > > > > > > I cannot get a dial tone from my USB FXS device. > > > > > > In my search through the list achives I found that it has been anissue with> > > some others. > > > > > > I ensured the modules where loaded in the right order. I managed toget> > > sound when I press the keys on the phone, but still no dialtone. > > > I used the astinstall script that came with my dev lite kit. (The FXOPCI> > > board is working fine) > > > > > > I tried unloading the modules and reloading them then ztcfg but stillno> > > dialtone. > > > > > > I am trying the install on a new box to elimitate the USB controlleras one> > > of the posibilities. > > > > > > I am using the sample configs. Perhaps I have not setup my extensions > > > properly. > > > > > > PII 350 64 meg ram redhat 9.0 dev lite kit. USB usb-ohci.c:usb-00:09.0,> > > OPTi Inc. 82C861 > > > > > > Can someone please help. > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
At 15:42 15-5-2003 -0500, you wrote:>Well your debug only shows that the phone didn't register itself with >asterisk. > >asterisk transmits: >"SIP/2.0 407 Proxy Authentication Required"Sorry Martin, the e-mail only showed the call-progress, not the registration. The registration process is below, and Asterisk does say the phone is registered and all... Any suggestions ? vectra*CLI> sip debug SIP Debugging Enabled Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249 Call-ID: 3350702873@172.28.4.182 CSeq: 1 REGISTER Contact: <sip:ata1-1@172.28.4.182:5060;transport=udp>;expires=3600 User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as3cef59dd Call-ID: 3350702873@172.28.4.182 CSeq: 1 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as3cef59dd Call-ID: 3350702873@172.28.4.182 CSeq: 1 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Proxy-Authenticate: Digest realm="asterisk", nonce="0fc9b3d4" Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249 Call-ID: 1635194524@172.28.4.182 CSeq: 1 REGISTER Contact: <sip:ata1-2@172.28.4.182:5060;transport=udp>;expires=3600 User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as29bb60e7 Call-ID: 1635194524@172.28.4.182 CSeq: 1 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as29bb60e7 Call-ID: 1635194524@172.28.4.182 CSeq: 1 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Proxy-Authenticate: Digest realm="asterisk", nonce="7b0bdc99" Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249 Call-ID: 3350702873@172.28.4.182 CSeq: 2 REGISTER Contact: * Expires: 0 User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as3cef59dd Call-ID: 3350702873@172.28.4.182 CSeq: 2 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as3cef59dd Call-ID: 3350702873@172.28.4.182 CSeq: 2 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Proxy-Authenticate: Digest realm="asterisk", nonce="2d8c795a" Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249 Call-ID: 1635194524@172.28.4.182 CSeq: 2 REGISTER Contact: * Expires: 0 User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as29bb60e7 Call-ID: 1635194524@172.28.4.182 CSeq: 2 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as29bb60e7 Call-ID: 1635194524@172.28.4.182 CSeq: 2 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Proxy-Authenticate: Digest realm="asterisk", nonce="49ef304a" Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249 Call-ID: 3350702873@172.28.4.182 CSeq: 3 REGISTER Contact: * Expires: 0 User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username="ata1-1",realm="asterisk",nonce="2d8c795a",uri="sip:217.114.97.249",response="369f5b517fdc46fd1be8e7157d7ef318" Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as3cef59dd Call-ID: 3350702873@172.28.4.182 CSeq: 3 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 -- Unregistered SIP 'ata1-1' Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as3cef59dd Call-ID: 3350702873@172.28.4.182 CSeq: 3 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Date: Fri, 16 May 2003 06:41:43 GMT Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249 Call-ID: 1635194524@172.28.4.182 CSeq: 3 REGISTER Contact: * Expires: 0 User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username="ata1-2",realm="asterisk",nonce="49ef304a",uri="sip:217.114.97.249",response="90a4d851ce0ac7547b0fda3ae49f51dd" Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as29bb60e7 Call-ID: 1635194524@172.28.4.182 CSeq: 3 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 -- Unregistered SIP 'ata1-2' Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as29bb60e7 Call-ID: 1635194524@172.28.4.182 CSeq: 3 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Date: Fri, 16 May 2003 06:41:43 GMT Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249 Call-ID: 3350702873@172.28.4.182 CSeq: 4 REGISTER Contact: <sip:ata1-1@130.89.224.0:5060;transport=udp>;expires=3600 User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as4380fb2e Call-ID: 3350702873@172.28.4.182 CSeq: 4 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as4380fb2e Call-ID: 3350702873@172.28.4.182 CSeq: 4 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Proxy-Authenticate: Digest realm="asterisk", nonce="24c97fd6" Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249 Call-ID: 1635194524@172.28.4.182 CSeq: 4 REGISTER Contact: <sip:ata1-2@130.89.224.0:5060;transport=udp>;expires=3600 User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as0bf2321b Call-ID: 1635194524@172.28.4.182 CSeq: 4 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as0bf2321b Call-ID: 1635194524@172.28.4.182 CSeq: 4 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Proxy-Authenticate: Digest realm="asterisk", nonce="3a7d93b1" Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249 Call-ID: 3350702873@172.28.4.182 CSeq: 5 REGISTER Contact: <sip:ata1-1@130.89.224.0:5060;transport=udp>;expires=3600 User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username="ata1-1",realm="asterisk",nonce="24c97fd6",uri="sip:217.114.97.249",response="84e454d8de99ac825bb810ebba63bd28" Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as4380fb2e Call-ID: 3350702873@172.28.4.182 CSeq: 5 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 -- Registered SIP 'ata1-1' at 130.89.224.240 port 35247 expires 3600 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as4380fb2e Call-ID: 3350702873@172.28.4.182 CSeq: 5 REGISTER User-Agent: Asterisk PBX Expires: 3600 Contact: <sip:ata1-1@217.114.97.249>;expires=3600 Date: Fri, 16 May 2003 06:41:44 GMT Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249 Call-ID: 1635194524@172.28.4.182 CSeq: 5 REGISTER Contact: <sip:ata1-2@130.89.224.0:5060;transport=udp>;expires=3600 User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username="ata1-2",realm="asterisk",nonce="3a7d93b1",uri="sip:217.114.97.249",response="cbd27b3a598d1f1ea1f83abc06a21bdb" Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as0bf2321b Call-ID: 1635194524@172.28.4.182 CSeq: 5 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 -- Registered SIP 'ata1-2' at 130.89.224.240 port 35247 expires 3600 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as0bf2321b Call-ID: 1635194524@172.28.4.182 CSeq: 5 REGISTER User-Agent: Asterisk PBX Expires: 3600 Contact: <sip:ata1-2@217.114.97.249>;expires=3600 Date: Fri, 16 May 2003 06:41:44 GMT Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249 Call-ID: 3350702873@172.28.4.182 CSeq: 6 REGISTER Contact: * Expires: 0 User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as4e60ad89 Call-ID: 3350702873@172.28.4.182 CSeq: 6 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as4e60ad89 Call-ID: 3350702873@172.28.4.182 CSeq: 6 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Proxy-Authenticate: Digest realm="asterisk", nonce="0ce91de4" Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249 Call-ID: 1635194524@172.28.4.182 CSeq: 6 REGISTER Contact: * Expires: 0 User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as4b895a75 Call-ID: 1635194524@172.28.4.182 CSeq: 6 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as4b895a75 Call-ID: 1635194524@172.28.4.182 CSeq: 6 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Proxy-Authenticate: Digest realm="asterisk", nonce="52ec29b4" Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249 Call-ID: 3350702873@172.28.4.182 CSeq: 7 REGISTER Contact: * Expires: 0 User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username="ata1-1",realm="asterisk",nonce="0ce91de4",uri="sip:217.114.97.249",response="622cfb574f5f9f5fbfc463f8e722901e" Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as4e60ad89 Call-ID: 3350702873@172.28.4.182 CSeq: 7 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 -- Unregistered SIP 'ata1-1' Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as4e60ad89 Call-ID: 3350702873@172.28.4.182 CSeq: 7 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Date: Fri, 16 May 2003 06:41:44 GMT Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249 Call-ID: 1635194524@172.28.4.182 CSeq: 7 REGISTER Contact: * Expires: 0 User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username="ata1-2",realm="asterisk",nonce="52ec29b4",uri="sip:217.114.97.249",response="c0a06a42e4876c2fa359464819998e24" Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as4b895a75 Call-ID: 1635194524@172.28.4.182 CSeq: 7 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 -- Unregistered SIP 'ata1-2' Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as4b895a75 Call-ID: 1635194524@172.28.4.182 CSeq: 7 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Date: Fri, 16 May 2003 06:41:44 GMT Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249 Call-ID: 3350702873@172.28.4.182 CSeq: 8 REGISTER Contact: <sip:ata1-1@130.89.224.0:5060;transport=udp>;expires=3600 User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as0b466ac3 Call-ID: 3350702873@172.28.4.182 CSeq: 8 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as0b466ac3 Call-ID: 3350702873@172.28.4.182 CSeq: 8 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Proxy-Authenticate: Digest realm="asterisk", nonce="7a33f45b" Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249 Call-ID: 1635194524@172.28.4.182 CSeq: 8 REGISTER Contact: <sip:ata1-2@130.89.224.0:5060;transport=udp>;expires=3600 User-Agent: Cisco ATA v2.15 ata18x (020927a) Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as3c64d8e9 Call-ID: 1635194524@172.28.4.182 CSeq: 8 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as3c64d8e9 Call-ID: 1635194524@172.28.4.182 CSeq: 8 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Proxy-Authenticate: Digest realm="asterisk", nonce="3fa3311f" Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249 Call-ID: 3350702873@172.28.4.182 CSeq: 9 REGISTER Contact: <sip:ata1-1@130.89.224.0:5060;transport=udp>;expires=3600 User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username="ata1-1",realm="asterisk",nonce="7a33f45b",uri="sip:217.114.97.249",response="655e6444df3e8d75810b80fd7098a9ca" Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as0b466ac3 Call-ID: 3350702873@172.28.4.182 CSeq: 9 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-1@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 -- Registered SIP 'ata1-1' at 130.89.224.240 port 35247 expires 3600 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-1@217.114.97.249 To: sip:ata1-1@217.114.97.249;tag=as0b466ac3 Call-ID: 3350702873@172.28.4.182 CSeq: 9 REGISTER User-Agent: Asterisk PBX Expires: 3600 Contact: <sip:ata1-1@217.114.97.249>;expires=3600 Date: Fri, 16 May 2003 06:41:44 GMT Content-Length: 0 to 130.89.224.240:35247 Sip read: > REGISTER sip:217.114.97.249 SIP/2.0 Via: SIP/2.0/UDP 172.28.4.182:5060 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249 Call-ID: 1635194524@172.28.4.182 CSeq: 9 REGISTER Contact: <sip:ata1-2@130.89.224.0:5060;transport=udp>;expires=3600 User-Agent: Cisco ATA v2.15 ata18x (020927a) Proxy-Authorization: Digest username="ata1-2",realm="asterisk",nonce="3fa3311f",uri="sip:217.114.97.249",response="7fc80b03860c9ff0c88f7765f56cd6c9" Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 172.28.4.182 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as3c64d8e9 Call-ID: 1635194524@172.28.4.182 CSeq: 9 REGISTER User-Agent: Asterisk PBX Contact: <sip:ata1-2@217.114.97.249> Content-Length: 0 to 130.89.224.240:35247 -- Registered SIP 'ata1-2' at 130.89.224.240 port 35247 expires 3600 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.4.182:5060;received=130.89.224.240:35247 From: sip:ata1-2@217.114.97.249 To: sip:ata1-2@217.114.97.249;tag=as3c64d8e9 Call-ID: 1635194524@172.28.4.182 CSeq: 9 REGISTER User-Agent: Asterisk PBX Expires: 3600 Contact: <sip:ata1-2@217.114.97.249>;expires=3600 Date: Fri, 16 May 2003 06:41:44 GMT Content-Length: 0 to 130.89.224.240:35247 vectra*CLI>
*talking to myself* My issues with the ATA are still unresolved. However, I am noticing things in the SIP dumps:>vectra*CLI> >Sip read: > >INVITE sip:0534280105@217.114.97.249;user=phone SIP/2.0 >Via: SIP/2.0/UDP 130.89.224.240:5060 >From: sip:ata1-1@217.114.97.249;tag=2733832243 >To: <sip:0534280105@217.114.97.249;user=phone> >Call-ID: 855024110@130.89.224.240 >CSeq: 1 INVITE >Contact: <sip:ata1-1@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 130.89.224.0Hmm, does my ATA request the sdp goes to 130.89.224.0 instead of 130.89.224.240 ? Did it read the Via header back wrong ?? Can this be causing the issues ? Simply setting a NATIP address makes it go wild and bang on the * server like a maniac :-(>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@217.114.97.249;tag=2733832243 >To: <sip:0534280105@217.114.97.249;user=phone>;tag=as7503b8bb >Call-ID: 855024110@130.89.224.240 >CSeq: 1 INVITE >User-Agent: Asterisk PBX >Contact: >Proxy-Authenticate: Digest realm="asterisk", nonce="1e537e10" >Content-Length: 0 > > > to 130.89.224.240:5060Florian
Whoa, I keep talking to myself. I've reset to factory settings, set it up again, this time with NATIP set and ConnectMode not 0x00460400 as was posted on the list some time back, but with 0x00060400 so the Via header should not matter now. Now it seems incoming calls work OK (after reinstating the portforward for RTP), but outbound is flaky, as the proxy authentication failed (even though this worked fine when registering to *!) Removing the secret from sip.conf solves this, and the phone works both ways! Now, why would proxy auth work when registering, but fail when calling ?? Florian