Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA.
Does anybody know what can be happing? Log is attached.. tks regards Oz
> 8 headers, 0 lines > Retransmitting #1 (NAT): > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 > From: sip:[EMAIL PROTECTED];tag=3346186142 > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as36ac1b92 > Call-ID: [EMAIL PROTECTED] > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Proxy-Authenticate: Digest realm="asterisk", nonce="4bd7a841" > Content-Length: 0 > > 290� > to 200.167.103.219:1025 > Sip read: LI> > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.150:5060 > From: sip:[EMAIL PROTECTED];tag=3346186142 > To: <sip:[EMAIL PROTECTED];user=phone> > Call-ID: [EMAIL PROTECTED] > CSeq: 2 INVITE > Contact: <sip:[EMAIL PROTECTED]:5060;transport=udp> > User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) > Proxy-Authorization: Digest > username="porto",realm="asterisk",nonce="4bd7a841",uri="sip:[EMAIL PROTECTED] > .77",response="1ecb99d4d5e23be179a9eb55eb33c62a" > Expires: 300 > Content-Length: 250 > Content-Type: application/sdp > > v=0 > o=porto 3642 3642 IN IP4 192.168.0.150 > s=ATA186 Call > c=IN IP4 192.168.0.150 > t=0 0 > m=audio 16384 RTP/AVP 18 8 0 101 > a=rtpmap:18 G729/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 12 headers, 11 lines > Using latest request as basis request > Sending to 192.168.0.150 : 5060 (NAT) > Found audio format UNKN > Found audio format ALAW > Found audio format UNKN > Found audio format UNKN > Found description format G729 > Found description format PCMA > Found description format PCMU > Found description format telephone-event > Capabilities: us - 256, them - 268/0, combined - 256 > Non-codec capabilities: us - 1, them - 1, combined - 1 > 10 headers, 0 lines > Reliably Transmitting: > OPTIONS sip:200.167.103.219:1025 SIP/2.0 > Via: SIP/2.0/UDP 200.170.156.77:5060;branch=z9hG4bK1937468f > From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as5566fcc8 > To: <sip:200.167.103.219:1025> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Content-Length: 0 > > (no NAT) to 200.167.103.219:1025 > Sip read: LI> > ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 > From: sip:[EMAIL PROTECTED];tag=3346186142 > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as36ac1b92 > Call-ID: [EMAIL PROTECTED] > CSeq: 1 ACK > User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) > Content-Length: 0 > > > 8 headers, 0 lines > Sip read: LI> > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.150:5060 > From: sip:[EMAIL PROTECTED];tag=3346186142 > To: <sip:[EMAIL PROTECTED];user=phone> > Call-ID: [EMAIL PROTECTED] > CSeq: 2 INVITE > Contact: <sip:[EMAIL PROTECTED]:5060;transport=udp> > User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) > Proxy-Authorization: Digest > username="porto",realm="asterisk",nonce="514a024a",uri="sip:[EMAIL PROTECTED] > .77",response="adb7da64c3f557d1db20b699c04f6d84" > Expires: 300 > Content-Length: 250 > Content-Type: application/sdp > > v=0 > o=porto 3692 3692 IN IP4 192.168.0.150 > s=ATA186 Call > c=IN IP4 192.168.0.150 > t=0 0 > m=audio 16384 RTP/AVP 18 8 0 101 > a=rtpmap:18 G729/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 12 headers, 11 lines > Using latest request as basis request > Sending to 192.168.0.150 : 5060 (non-NAT) > Found audio format UNKN > Found audio format ALAW > Found audio format UNKN > Found audio format UNKN > Found description format G729 > Found description format PCMA > Found description format PCMU > Found description format telephone-event > Capabilities: us - 256, them - 268/0, combined - 256 > Non-codec capabilities: us - 1, them - 1, combined - 1 > Reliably Transmitting (NAT): > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 > From: sip:[EMAIL PROTECTED];tag=3346186142 > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as046b1041 > Call-ID: [EMAIL PROTECTED] > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Proxy-Authenticate: Digest realm="asterisk", nonce="6512ffab" > Content-Length: 0 > > > to 200.167.103.219:1025 > Sip read: LI> > ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 > From: sip:[EMAIL PROTECTED];tag=3346186142 > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as36ac1b92 > Call-ID: [EMAIL PROTECTED] > CSeq: 1 ACK > User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) > Content-Length: 0 > > > 8 headers, 0 lines > Retransmitting #1 (no NAT): > OPTIONS sip:200.167.103.219:1025 SIP/2.0 > Via: SIP/2.0/UDP 200.170.156.77:5060;branch=z9hG4bK1937468f > From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as5566fcc8 > To: <sip:200.167.103.219:1025> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Content-Length: 0 > > tent- > to 200.167.103.219:1025 > Sip read: LI> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 200.170.156.77:5060;branch=z9hG4bK1937468f > From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as5566fcc8 > To: <sip:200.167.103.219:1025>;tag=3346186142 > Call-ID: [EMAIL PROTECTED] > CSeq: 102 OPTIONS > Server: Cisco ATA 186 v2.16.1 ata18x (030709a) > Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER > Content-Length: 250 > Content-Type: application/sdp > > v=0 > o=porto 3779 3779 IN IP4 192.168.0.150 > s=ATA186 Call > c=IN IP4 192.168.0.150 > t=0 0 > m=audio 16384 RTP/AVP 0 8 18 101 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:18 G729/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 10 headers, 11 lines > Retransmitting #1 (NAT): > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 > From: sip:[EMAIL PROTECTED];tag=3346186142 > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as046b1041 > Call-ID: [EMAIL PROTECTED] > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Proxy-Authenticate: Digest realm="asterisk", nonce="6512ffab" > Content-Length: 0 > > > to 200.167.103.219:1025 > Sip read: LI> > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.150:5060 > From: sip:[EMAIL PROTECTED];tag=3346186142 > To: <sip:[EMAIL PROTECTED];user=phone> > Call-ID: [EMAIL PROTECTED] > CSeq: 2 INVITE > Contact: <sip:[EMAIL PROTECTED]:5060;transport=udp> > User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) > Proxy-Authorization: Digest > username="porto",realm="asterisk",nonce="4bd7a841",uri="sip:[EMAIL PROTECTED] > .77",response="1ecb99d4d5e23be179a9eb55eb33c62a" > Expires: 300 > Content-Length: 250 > Content-Type: application/sdp > > v=0 > o=porto 3792 3792 IN IP4 192.168.0.150 > s=ATA186 Call > c=IN IP4 192.168.0.150 > t=0 0 > m=audio 16384 RTP/AVP 18 8 0 101 > a=rtpmap:18 G729/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 12 headers, 11 lines > Ignoring this request > Retransmitting #2 (NAT): > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 > From: sip:[EMAIL PROTECTED];tag=3346186142 > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as046b1041 > Call-ID: [EMAIL PROTECTED] > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Proxy-Authenticate: Digest realm="asterisk", nonce="6512ffab" > Content-Length: 0 > > > to 200.167.103.219:1025 > Sip read: LI> > ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 > From: sip:[EMAIL PROTECTED];tag=3346186142 > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as046b1041 > Call-ID: [EMAIL PROTECTED] > CSeq: 2 ACK > User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) > Content-Length: 0 > > > 8 headers, 0 lines > Sip read: LI> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 200.170.156.77:5060;branch=z9hG4bK1937468f > From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as5566fcc8 > To: <sip:200.167.103.219:1025>;tag=3346186142 > Call-ID: [EMAIL PROTECTED] > CSeq: 102 OPTIONS > Server: Cisco ATA 186 v2.16.1 ata18x (030709a) > Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER > Content-Length: 250 > Content-Type: application/sdp > > v=0 > o=porto 3885 3885 IN IP4 192.168.0.150 > s=ATA186 Call > c=IN IP4 192.168.0.150 > t=0 0 > m=audio 16384 RTP/AVP 0 8 18 101 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:18 G729/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15
