[Asterisk-Users] ATA call

Tue Jan 6 15:41:35 MST 2004

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
-------------- next part --------------
> 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:porto at 200.170.156.77;tag=3346186142
> To: <sip:4057 at 200.170.156.77;user=phone>;tag=as36ac1b92
> Call-ID: 471622025 at 192.168.0.150
> 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
> 
> 290Ñ
>  to 200.167.103.219:1025
> Sip read: LI>
> INVITE sip:4057 at 200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060
> From: sip:porto at 200.170.156.77;tag=3346186142
> To: <sip:4057 at 200.170.156.77;user=phone>
> Call-ID: 471622025 at 192.168.0.150
> CSeq: 2 INVITE
> Contact: <sip:porto at 192.168.0.150:5060;transport=udp>
> User-Agent: Cisco ATA 186  v2.16.1 ata18x (030709a)
> Proxy-Authorization: Digest
> username="porto",realm="asterisk",nonce="4bd7a841",uri="sip:4057 at 200.170.156
> .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:asterisk at 200.170.156.77>;tag=as5566fcc8
> To: <sip:200.167.103.219:1025>
> Contact: <sip:asterisk at 200.170.156.77>
> Call-ID: 2d2e8097757b385a3444a1ab239ab86c at 200.170.156.77
> 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:4057 at 200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219
> From: sip:porto at 200.170.156.77;tag=3346186142
> To: <sip:4057 at 200.170.156.77;user=phone>;tag=as36ac1b92
> Call-ID: 471622025 at 192.168.0.150
> 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:4057 at 200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060
> From: sip:porto at 200.170.156.77;tag=3346186142
> To: <sip:4057 at 200.170.156.77;user=phone>
> Call-ID: 471622025 at 192.168.0.150
> CSeq: 2 INVITE
> Contact: <sip:porto at 192.168.0.150:5060;transport=udp>
> User-Agent: Cisco ATA 186  v2.16.1 ata18x (030709a)
> Proxy-Authorization: Digest
> username="porto",realm="asterisk",nonce="514a024a",uri="sip:4057 at 200.170.156
> .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:porto at 200.170.156.77;tag=3346186142
> To: <sip:4057 at 200.170.156.77;user=phone>;tag=as046b1041
> Call-ID: 471622025 at 192.168.0.150
> 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:4057 at 200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219
> From: sip:porto at 200.170.156.77;tag=3346186142
> To: <sip:4057 at 200.170.156.77;user=phone>;tag=as36ac1b92
> Call-ID: 471622025 at 192.168.0.150
> 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:asterisk at 200.170.156.77>;tag=as5566fcc8
> To: <sip:200.167.103.219:1025>
> Contact: <sip:asterisk at 200.170.156.77>
> Call-ID: 2d2e8097757b385a3444a1ab239ab86c at 200.170.156.77
> 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:asterisk at 200.170.156.77>;tag=as5566fcc8
> To: <sip:200.167.103.219:1025>;tag=3346186142
> Call-ID: 2d2e8097757b385a3444a1ab239ab86c at 200.170.156.77
> 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:porto at 200.170.156.77;tag=3346186142
> To: <sip:4057 at 200.170.156.77;user=phone>;tag=as046b1041
> Call-ID: 471622025 at 192.168.0.150
> 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:4057 at 200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060
> From: sip:porto at 200.170.156.77;tag=3346186142
> To: <sip:4057 at 200.170.156.77;user=phone>
> Call-ID: 471622025 at 192.168.0.150
> CSeq: 2 INVITE
> Contact: <sip:porto at 192.168.0.150:5060;transport=udp>
> User-Agent: Cisco ATA 186  v2.16.1 ata18x (030709a)
> Proxy-Authorization: Digest
> username="porto",realm="asterisk",nonce="4bd7a841",uri="sip:4057 at 200.170.156
> .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:porto at 200.170.156.77;tag=3346186142
> To: <sip:4057 at 200.170.156.77;user=phone>;tag=as046b1041
> Call-ID: 471622025 at 192.168.0.150
> 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:4057 at 200.170.156.77;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219
> From: sip:porto at 200.170.156.77;tag=3346186142
> To: <sip:4057 at 200.170.156.77;user=phone>;tag=as046b1041
> Call-ID: 471622025 at 192.168.0.150
> 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:asterisk at 200.170.156.77>;tag=as5566fcc8
> To: <sip:200.167.103.219:1025>;tag=3346186142
> Call-ID: 2d2e8097757b385a3444a1ab239ab86c at 200.170.156.77
> 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