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- [root@localhost ~]# asterisk -r
- Asterisk 11.25.3, Copyright (C) 1999 - 2013 Digium, Inc. and others.
- Created by Mark Spencer <markster@digium.com>
- Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail s.
- This is free software, with components licensed under the GNU General Public
- License version 2 and other licenses; you are welcome to redistribute it under
- certain conditions. Type 'core show license' for details.
- =========================================================================
- Connected to Asterisk 11.25.3 currently running on localhost (pid = 2499)
- localhost*CLI> SIP SET DEBUG peer tk_saida
- SIP Debugging Enabled for IP: 200.184.77.169
- Audio is at 16702
- Adding codec 100004 (alaw) to SDP
- Adding codec 100003 (ulaw) to SDP
- Adding codec 100008 (g729) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 200.184.77.169:5060:
- INVITE sip:33513872@200.184.77.169 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK42762661;rport
- Max-Forwards: 70
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>
- Contact: <sip:4733513872@192.168.1.111:5060>
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 102 INVITE
- User-Agent: FPBX-2.11.0(11.25.3)
- Date: Mon, 30 Apr 2018 14:42:55 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 309
- v=0
- o=root 1331459369 1331459369 IN IP4 192.168.1.111
- s=Asterisk PBX 11.25.3
- c=IN IP4 192.168.1.111
- t=0 0
- m=audio 16702 RTP/AVP 8 0 18 101
- a=rtpmap:8 PCMA/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:18 G729/8000
- a=fmtp:18 annexb=no
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- <--- SIP read from UDP:200.184.77.169:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK42762661;received=187.85.174.1 34;rport=5060
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>;tag=as6112cc95
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 102 INVITE
- User-Agent: InforSOLutions Billing
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f043f66"
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Transmitting (NAT) to 200.184.77.169:5060:
- ACK sip:33513872@200.184.77.169 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK42762661;rport
- Max-Forwards: 70
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>;tag=as6112cc95
- Contact: <sip:4733513872@192.168.1.111:5060>
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 102 ACK
- User-Agent: FPBX-2.11.0(11.25.3)
- Content-Length: 0
- ---
- Audio is at 16702
- Adding codec 100004 (alaw) to SDP
- Adding codec 100003 (ulaw) to SDP
- Adding codec 100008 (g729) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 200.184.77.169:5060:
- INVITE sip:33513872@200.184.77.169 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;rport
- Max-Forwards: 70
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>
- Contact: <sip:4733513872@192.168.1.111:5060>
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 103 INVITE
- User-Agent: FPBX-2.11.0(11.25.3)
- Proxy-Authorization: Digest username="5076", realm="asterisk", algorithm=MD5, ur i="sip:33513872@200.184.77.169", nonce="2f043f66", response="f5299ae6e1cba9068ff 8d67f63b00add"
- Date: Mon, 30 Apr 2018 14:42:55 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 309
- v=0
- o=root 1331459369 1331459370 IN IP4 192.168.1.111
- s=Asterisk PBX 11.25.3
- c=IN IP4 192.168.1.111
- t=0 0
- m=audio 16702 RTP/AVP 8 0 18 101
- a=rtpmap:8 PCMA/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:18 G729/8000
- a=fmtp:18 annexb=no
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- <--- SIP read from UDP:200.184.77.169:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;received=187.85.174.1 34;rport=5060
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 103 INVITE
- User-Agent: InforSOLutions Billing
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:33513872@200.184.77.169>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- [2018-04-30 11:42:59] WARNING[6332][C-00000040]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
- <--- SIP read from UDP:200.184.77.169:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;received=187.85.174.1 34;rport=5060
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>;tag=as4b2a9893
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 103 INVITE
- User-Agent: InforSOLutions Billing
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:33513872@200.184.77.169>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- list_route: hop: <sip:33513872@200.184.77.169>
- <--- SIP read from UDP:200.184.77.169:5060 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;received=187.85.174.1 34;rport=5060
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>;tag=as4b2a9893
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 103 INVITE
- User-Agent: InforSOLutions Billing
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:33513872@200.184.77.169>
- Content-Type: application/sdp
- Content-Length: 313
- v=0
- o=root 3360 3360 IN IP4 200.184.77.169
- s=session
- c=IN IP4 200.184.77.169
- t=0 0
- m=audio 16820 RTP/AVP 18 8 0 101
- a=rtpmap:18 G729/8000
- a=fmtp:18 annexb=no
- a=rtpmap:8 PCMA/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------->
- --- (12 headers 15 lines) ---
- list_route: hop: <sip:33513872@200.184.77.169>
- Found RTP audio format 18
- Found RTP audio format 8
- Found RTP audio format 0
- Found RTP audio format 101
- Found audio description format G729 for ID 18
- Found audio description format PCMA for ID 8
- Found audio description format PCMU for ID 0
- Found audio description format telephone-event for ID 101
- Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothin g)/text=(nothing), combined - (ulaw|alaw|g729)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 200.184.77.169:16820
- Scheduling destruction of SIP dialog '660c74301af7b2086d94e94261191fc3@192.168.1 .111:5060' in 6400 ms (Method: INVITE)
- Reliably Transmitting (NAT) to 200.184.77.169:5060:
- CANCEL sip:33513872@200.184.77.169 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;rport
- Max-Forwards: 70
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 103 CANCEL
- User-Agent: FPBX-2.11.0(11.25.3)
- Content-Length: 0
- ---
- Scheduling destruction of SIP dialog '660c74301af7b2086d94e94261191fc3@192.168.1 .111:5060' in 6400 ms (Method: INVITE)
- <--- SIP read from UDP:200.184.77.169:5060 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;received=187.85.174.1 34;rport=5060
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>;tag=as4b2a9893
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 103 INVITE
- User-Agent: InforSOLutions Billing
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- Transmitting (NAT) to 200.184.77.169:5060:
- ACK sip:33513872@200.184.77.169 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;rport
- Max-Forwards: 70
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>;tag=as4b2a9893
- Contact: <sip:4733513872@192.168.1.111:5060>
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 103 ACK
- User-Agent: FPBX-2.11.0(11.25.3)
- Content-Length: 0
- ---
- Scheduling destruction of SIP dialog '660c74301af7b2086d94e94261191fc3@192.168.1 .111:5060' in 6400 ms (Method: INVITE)
- Retransmitting #1 (NAT) to 200.184.77.169:5060:
- CANCEL sip:33513872@200.184.77.169 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;rport
- Max-Forwards: 70
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 103 CANCEL
- User-Agent: FPBX-2.11.0(11.25.3)
- Content-Length: 0
- ---
- <--- SIP read from UDP:200.184.77.169:5060 --->
- SIP/2.0 481 Call leg/transaction does not exist
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5dad6e92;received=187.85.174.1 34;rport=5060
- From: <sip:4733513872@192.168.1.111>;tag=as20b1d26c
- To: <sip:33513872@200.184.77.169>;tag=as709ee3d5
- Call-ID: 660c74301af7b2086d94e94261191fc3@192.168.1.111:5060
- CSeq: 103 CANCEL
- User-Agent: InforSOLutions Billing
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- [2018-04-30 11:43:03] WARNING[2586][C-0000003f]: chan_sip.c:24287 handle_respons e: Remote host can't match request CANCEL to call '660c74301af7b2086d94e94261191 fc3@192.168.1.111:5060'. Giving up.
- Really destroying SIP dialog '7efab3fa4587c870636031990fe0d375@200.184.77.169' M ethod: OPTIONS
- Really destroying SIP dialog '660c74301af7b2086d94e94261191fc3@192.168.1.111:506 0' Method: INVITE
- Audio is at 12698
- Adding codec 100004 (alaw) to SDP
- Adding codec 100003 (ulaw) to SDP
- Adding codec 100008 (g729) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 200.184.77.169:5060:
- INVITE sip:33555748@200.184.77.169 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK4469cb93;rport
- Max-Forwards: 70
- From: <sip:4733513872@192.168.1.111>;tag=as2636fbad
- To: <sip:33555748@200.184.77.169>
- Contact: <sip:4733513872@192.168.1.111:5060>
- Call-ID: 40b221be1368da20765d11de78d950e0@192.168.1.111:5060
- CSeq: 102 INVITE
- User-Agent: FPBX-2.11.0(11.25.3)
- Date: Mon, 30 Apr 2018 14:43:13 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 307
- v=0
- o=root 878878280 878878280 IN IP4 192.168.1.111
- s=Asterisk PBX 11.25.3
- c=IN IP4 192.168.1.111
- t=0 0
- m=audio 12698 RTP/AVP 8 0 18 101
- a=rtpmap:8 PCMA/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:18 G729/8000
- a=fmtp:18 annexb=no
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- <--- SIP read from UDP:200.184.77.169:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK4469cb93;received=187.85.174.1 34;rport=5060
- From: <sip:4733513872@192.168.1.111>;tag=as2636fbad
- To: <sip:33555748@200.184.77.169>
- Call-ID: 40b221be1368da20765d11de78d950e0@192.168.1.111:5060
- CSeq: 102 INVITE
- User-Agent: InforSOLutions Billing
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:33555748@200.184.77.169>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- <--- SIP read from UDP:200.184.77.169:5060 --->
- SIP/2.0 603 Declined
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK4469cb93;received=187.85.174.1 34;rport=5060
- From: <sip:4733513872@192.168.1.111>;tag=as2636fbad
- To: <sip:33555748@200.184.77.169>;tag=as4827d7e2
- Call-ID: 40b221be1368da20765d11de78d950e0@192.168.1.111:5060
- CSeq: 102 INVITE
- User-Agent: InforSOLutions Billing
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- Transmitting (NAT) to 200.184.77.169:5060:
- ACK sip:33555748@200.184.77.169 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK4469cb93;rport
- Max-Forwards: 70
- From: <sip:4733513872@192.168.1.111>;tag=as2636fbad
- To: <sip:33555748@200.184.77.169>;tag=as4827d7e2
- Contact: <sip:4733513872@192.168.1.111:5060>
- Call-ID: 40b221be1368da20765d11de78d950e0@192.168.1.111:5060
- CSeq: 102 ACK
- User-Agent: FPBX-2.11.0(11.25.3)
- Content-Length: 0
- ---
- Really destroying SIP dialog '40b221be1368da20765d11de78d950e0@192.168.1.111:506 0' Method: INVITE
- [2018-04-30 11:43:18] NOTICE[2586]: chan_sip.c:16725 check_auth: Correct auth, b ut based on stale nonce received from '"204" <sip:204@192.168.1.111>;tag=2789253 13'
- [2018-04-30 11:43:18] NOTICE[2586]: chan_sip.c:16725 check_auth: Correct auth, b ut based on stale nonce received from '"47 99985 9396"<sip:210@192.168.1.111:506 0;transport=UDP>;tag=88594b00'
- [2018-04-30 11:43:19] NOTICE[2586]: chan_sip.c:16725 check_auth: Correct auth, b ut based on stale nonce received from '"47 99985 9396"<sip:210@192.168.1.111:506 0;transport=UDP>;tag=88594b00'
- localhost*CLI>
- Disconnected from Asterisk server
- Asterisk cleanly ending (0).
- Executing last minute cleanups
- [root@localhost ~]#
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