Advertisement
Not a member of Pastebin yet?
Sign Up,
it unlocks many cool features!
- localhost*CLI>
- localhost*CLI>
- <--- SIP read from UDP://86.53.0.135:5060 --->
- <------------->
- localhost*CLI>
- <--- SIP read from UDP://64.154.41.150:5060 --->
- INVITE sip:5854928868@68.44.81.246 SIP/2.0
- Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK2bf712bf;rport
- From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
- To: <sip:5854928868@68.44.81.246>
- Contact: <sip:6465735875@64.154.41.150>
- Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Date: Thu, 08 Apr 2010 02:14:51 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
- Supported: replaces
- Content-Type: application/sdp
- Content-Length: 311
- v=0
- o=root 3339 3339 IN IP4 64.154.41.150
- s=session
- c=IN IP4 64.154.41.150
- t=0 0
- m=audio 15118 RTP/AVP 0 8 18 100
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:18 G729/8000
- a=fmtp:18 annexb=no
- a=rtpmap:100 telephone-event/8000
- a=fmtp:100 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------->
- --- (14 headers 15 lines) ---
- == Using SIP RTP CoS mark 5
- Sending to 64.154.41.150 : 5060 (no NAT)
- Using INVITE request as basis request - 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
- No user '6465735875' in SIP users list
- Found peer 'ipcom_incoming' for '6465735875' from 64.154.41.150:5060
- Found RTP audio format 0
- Found RTP audio format 8
- Found RTP audio format 18
- Found RTP audio format 100
- Found audio description format PCMU for ID 0
- Found audio description format PCMA for ID 8
- Found audio description format G729 for ID 18
- Found audio description format telephone-event for ID 100
- Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- Peer audio RTP is at port 64.154.41.150:15118
- Looking for 5854928868 in incoming_calls (domain 68.44.81.246)
- list_route: hop: <sip:6465735875@64.154.41.150>
- <--- Transmitting (no NAT) to 64.154.41.150:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK2bf712bf;received=64.154.41.150;rport=5060
- From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
- To: <sip:5854928868@68.44.81.246>
- Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.6.0.26
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:5854928868@192.168.1.158>
- Content-Length: 0
- <------------>
- -- Executing [5854928868@incoming_calls:1] GotoIf("SIP/ipcom_incoming-0000000a", "1?disa") in new stack
- -- Goto (incoming_calls,5854928868,3)
- -- Executing [5854928868@incoming_calls:3] Answer("SIP/ipcom_incoming-0000000a", "") in new stack
- Audio is at 192.168.1.158 port 18332
- Adding codec 0x4 (ulaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- <--- Reliably Transmitting (no NAT) to 64.154.41.150:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK2bf712bf;received=64.154.41.150;rport=5060
- From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
- To: <sip:5854928868@68.44.81.246>;tag=as38830dd8
- Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.6.0.26
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:5854928868@192.168.1.158>
- Content-Type: application/sdp
- Content-Length: 266
- v=0
- o=root 1978871929 1978871929 IN IP4 192.168.1.158
- s=Asterisk PBX 1.6.0.26
- c=IN IP4 192.168.1.158
- t=0 0
- m=audio 18332 RTP/AVP 0 100
- a=rtpmap:0 PCMU/8000
- a=rtpmap:100 telephone-event/8000
- a=fmtp:100 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------>
- localhost*CLI>
- <--- SIP read from UDP://64.154.41.150:5060 --->
- ACK sip:5854928868@192.168.1.158 SIP/2.0
- Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK17818e22;rport
- From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
- To: <sip:5854928868@68.44.81.246>;tag=as38830dd8
- Contact: <sip:6465735875@64.154.41.150>
- Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
- CSeq: 102 ACK
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- -- Executing [5854928868@incoming_calls:4] Set("SIP/ipcom_incoming-0000000a", "TIMEOUT(digit)=5") in new stack
- -- Digit timeout set to 5
- -- Executing [5854928868@incoming_calls:5] Set("SIP/ipcom_incoming-0000000a", "TIMEOUT(response)=9") in new stack
- -- Response timeout set to 9
- -- Executing [5854928868@incoming_calls:6] Authenticate("SIP/ipcom_incoming-0000000a", "123") in new stack
- -- <SIP/ipcom_incoming-0000000a> Playing 'agent-pass.gsm' (language 'en')
- -- <SIP/ipcom_incoming-0000000a> Playing 'auth-thankyou.gsm' (language 'en')
- -- Executing [5854928868@incoming_calls:7] DISA("SIP/ipcom_incoming-0000000a", "no-password,outcontext") in new stack
- -- Executing [01117327896169@outcontext:1] Dial("SIP/ipcom_incoming-0000000a", "SIP/ipc_outgoing/01117327896169") in new stack
- == Using SIP RTP CoS mark 5
- Audio is at 192.168.1.158 port 15464
- Adding codec 0x4 (ulaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (no NAT) to 86.53.0.135:5060:
- INVITE sip:01117327896169@sipagate.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bK03e07a7a;rport
- Max-Forwards: 70
- From: "6465735875" <sip:56000351@sipagate.com>;tag=as3fa6d5bc
- To: <sip:01117327896169@sipagate.com>
- Contact: <sip:56000351@192.168.1.158>
- Call-ID: 3f9b887d306591ca74fdbf8b18c203ca@sipagate.com
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.6.0.26
- Date: Thu, 08 Apr 2010 02:15:10 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 266
- v=0
- o=root 1266698409 1266698409 IN IP4 192.168.1.158
- s=Asterisk PBX 1.6.0.26
- c=IN IP4 192.168.1.158
- t=0 0
- m=audio 15464 RTP/AVP 0 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- ---
- -- Called ipc_outgoing/01117327896169
- localhost*CLI>
- <--- SIP read from UDP://86.53.0.135:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bK03e07a7a;rport=1025;received=68.44.81.246
- From: "6465735875" <sip:56000351@sipagate.com>;tag=as3fa6d5bc
- To: <sip:01117327896169@sipagate.com>;tag=4938594e7c94d95c79aee0bf2e224375.96f7
- Call-ID: 3f9b887d306591ca74fdbf8b18c203ca@sipagate.com
- CSeq: 102 INVITE
- WWW-Authenticate: Digest realm="sipagate.com", nonce="4bbd3a42a324c5cd2241544ee91217a49f6ac865"
- Server: Sip EXpress router (0.9.6 (i386/linux))
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- Transmitting (no NAT) to 86.53.0.135:5060:
- ACK sip:01117327896169@sipagate.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bK03e07a7a;rport
- Max-Forwards: 70
- From: "6465735875" <sip:56000351@sipagate.com>;tag=as3fa6d5bc
- To: <sip:01117327896169@sipagate.com>;tag=4938594e7c94d95c79aee0bf2e224375.96f7
- Contact: <sip:56000351@192.168.1.158>
- Call-ID: 3f9b887d306591ca74fdbf8b18c203ca@sipagate.com
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 1.6.0.26
- Content-Length: 0
- ---
- [Apr 7 22:15:10] NOTICE[22058]: chan_sip.c:16131 handle_response_invite: Failed to authenticate on INVITE to '"6465735875" <sip:56000351@sipagate.com>;tag=as3fa6d5bc'
- -- SIP/ipc_outgoing-0000000b is circuit-busy
- == Everyone is busy/congested at this time (1:0/1/0)
- -- Auto fallthrough, channel 'SIP/ipcom_incoming-0000000a' status is 'CONGESTION'
- Really destroying SIP dialog '3f9b887d306591ca74fdbf8b18c203ca@sipagate.com' Method: INVITE
- Really destroying SIP dialog '1ef7f81d0e715cdb063e1a564a9573dd@127.0.0.1' Method: REGISTER
- Really destroying SIP dialog '27a2531352c461a258534e476fff0605@127.0.0.1' Method: REGISTER
- localhost*CLI>
- <--- SIP read from UDP://64.154.41.150:5060 --->
- BYE sip:5854928868@192.168.1.158 SIP/2.0
- Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK091b07b8;rport
- From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
- To: <sip:5854928868@68.44.81.246>;tag=as38830dd8
- Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
- CSeq: 103 BYE
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- X-Asterisk-HangupCause: Normal Clearing
- X-Asterisk-HangupCauseCode: 16
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Sending to 64.154.41.150 : 5060 (no NAT)
- <--- Transmitting (no NAT) to 64.154.41.150:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 64.154.41.150:5060;branch=z9hG4bK091b07b8;received=64.154.41.150;rport=5060
- From: "6465735875" <sip:6465735875@64.154.41.150>;tag=as2ffd5269
- To: <sip:5854928868@68.44.81.246>;tag=as38830dd8
- Call-ID: 457a25bd233a60d9020e31ff3b925e2c@64.154.41.150
- CSeq: 103 BYE
- User-Agent: Asterisk PBX 1.6.0.26
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- localhost*CLI>
- <--- SIP read from UDP://86.53.0.135:5060 --->
- <------------->
- Really destroying SIP dialog '457a25bd233a60d9020e31ff3b925e2c@64.154.41.150' Method: BYE
- localhost*CLI>
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement