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- Asterisk 1.4.26.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
- Created by Mark Spencer <markster@digium.com>
- Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
- 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 1.4.26.1 currently running on meetmecall-01 (pid = 2685)
- meetmecall-01*CLI>
- Core debug is at least 10
- [Kmeetmecall-01*CLI>
- [May 17 20:04:06] NOTICE[2708]: chan_sip.c:7753 sip_reregister: -- Re-registration for 1429999999@voip1.sig.itsp.nl
- REGISTER 13 headers, 0 lines
- REGISTER attempt 1 to 1429999999@voip1.sig.itsp.nl
- Reliably Transmitting (no NAT) to xx.xxx.204.68:5060:
- REGISTER sip:voip1.sig.itsp.nl SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK590a406b;rport
- From: <sip:1429999999@voip1.sig.itsp.nl>;tag=as2bcd00dc
- To: <sip:1429999999@voip1.sig.itsp.nl>
- Call-ID: 7f34e9117e68dad470385cab45a5e064@xx.xx.241.180
- CSeq: 160 REGISTER
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Authorization: Digest username="1429999999", realm="itsp.nl", algorithm=MD5, uri="sip:voip1.sig.itsp.nl", nonce="3929cf94", response="bb5acb40a340de5f14bcc63f985d3051"
- Expires: 120
- Contact: <sip:s@xx.xx.241.180>
- Event: registration
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.204.68:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK590a406b;received=xx.xx.241.180;rport=5060
- From: <sip:1429999999@voip1.sig.itsp.nl>;tag=as2bcd00dc
- To: <sip:1429999999@voip1.sig.itsp.nl>
- Call-ID: 7f34e9117e68dad470385cab45a5e064@xx.xx.241.180
- CSeq: 160 REGISTER
- User-Agent: itsp VoIP Interconnect
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
- Supported: replaces
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.204.68:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK590a406b;received=xx.xx.241.180;rport=5060
- From: <sip:1429999999@voip1.sig.itsp.nl>;tag=as2bcd00dc
- To: <sip:1429999999@voip1.sig.itsp.nl>;tag=as343649a4
- Call-ID: 7f34e9117e68dad470385cab45a5e064@xx.xx.241.180
- CSeq: 160 REGISTER
- User-Agent: itsp VoIP Interconnect
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
- Supported: replaces
- WWW-Authenticate: Digest algorithm=MD5, realm="itsp.nl", nonce="4b514668"
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- [Kmeetmecall-01*CLI>
- Responding to challenge, registration to domain/host name voip1.sig.itsp.nl
- [Kmeetmecall-01*CLI>
- REGISTER 13 headers, 0 lines
- REGISTER attempt 2 to 1429999999@voip1.sig.itsp.nl
- [Kmeetmecall-01*CLI>
- Reliably Transmitting (no NAT) to xx.xxx.204.68:5060:
- REGISTER sip:voip1.sig.itsp.nl SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK7317fc04;rport
- From: <sip:1429999999@voip1.sig.itsp.nl>;tag=as280729a5
- To: <sip:1429999999@voip1.sig.itsp.nl>
- Call-ID: 7f34e9117e68dad470385cab45a5e064@xx.xx.241.180
- CSeq: 161 REGISTER
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Authorization: Digest username="1429999999", realm="itsp.nl", algorithm=MD5, uri="sip:voip1.sig.itsp.nl", nonce="4b514668", response="5bbce326e107aed7d02d4b3ed205d421"
- Expires: 120
- Contact: <sip:s@xx.xx.241.180>
- Event: registration
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.204.68:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK7317fc04;received=xx.xx.241.180;rport=5060
- From: <sip:1429999999@voip1.sig.itsp.nl>;tag=as280729a5
- To: <sip:1429999999@voip1.sig.itsp.nl>
- Call-ID: 7f34e9117e68dad470385cab45a5e064@xx.xx.241.180
- CSeq: 161 REGISTER
- User-Agent: itsp VoIP Interconnect
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
- Supported: replaces
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.204.68:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK7317fc04;received=xx.xx.241.180;rport=5060
- From: <sip:1429999999@voip1.sig.itsp.nl>;tag=as280729a5
- To: <sip:1429999999@voip1.sig.itsp.nl>;tag=as343649a4
- Call-ID: 7f34e9117e68dad470385cab45a5e064@xx.xx.241.180
- CSeq: 161 REGISTER
- User-Agent: itsp VoIP Interconnect
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
- Supported: replaces
- Expires: 120
- Contact: <sip:s@xx.xx.241.180>;expires=120
- Date: Mon, 17 May 2010 18:02:48 GMT
- Content-Length: 0
- <------------->
- --- (13 headers 0 lines) ---
- [Kmeetmecall-01*CLI>
- Scheduling destruction of SIP dialog '7f34e9117e68dad470385cab45a5e064@xx.xx.241.180' in 32000 ms (Method: REGISTER)
- [Kmeetmecall-01*CLI>
- [May 17 20:04:06] NOTICE[2708]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for voip1.sig.itsp.nl is 120 sec (Scheduling reregistration in 105 s)
- [Kmeetmecall-01*CLI>
- Really destroying SIP dialog '7392a5a40317eb05014f99d419bbf5c9@xx.xx.241.180' Method: REGISTER
- [Kmeetmecall-01*CLI>
- Really destroying SIP dialog '2175985755b07ecd1cd804db4a708021@xx.xx.241.180' Method: REGISTER
- [Kmeetmecall-01*CLI>
- Reliably Transmitting (NAT) to xx.xx.71.170:1030:
- OPTIONS sip:505@xx.xx.71.170:5060 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK2653b67f;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as07a56e9e
- To: <sip:505@xx.xx.71.170:5060>
- Contact: <sip:asterisk@xx.xx.241.180>
- Call-ID: 7c2ceddf5cb6e0152eb7e67e74701cc9@xx.xx.241.180
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Date: Mon, 17 May 2010 18:04:11 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xx.71.170:1030 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK2653b67f;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as07a56e9e
- To: <sip:505@xx.xx.71.170:5060>;tag=602B828B-603E4C74
- CSeq: 102 OPTIONS
- Call-ID: 7c2ceddf5cb6e0152eb7e67e74701cc9@xx.xx.241.180
- Contact: <sip:505@xx.xx.71.170:5060>
- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
- User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049
- [Kmeetmecall-01*CLI>
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- [Kmeetmecall-01*CLI>
- Really destroying SIP dialog '7c2ceddf5cb6e0152eb7e67e74701cc9@xx.xx.241.180' Method: OPTIONS
- [Kmeetmecall-01*CLI>
- Reliably Transmitting (NAT) to xx.xx.71.170:5060:
- OPTIONS sip:504@192.168.1.111:5060 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK04978a6e;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as773a250d
- To: <sip:504@192.168.1.111:5060>
- Contact: <sip:asterisk@xx.xx.241.180>
- Call-ID: 4658ee29294db8e30c6816d2609518ad@xx.xx.241.180
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Date: Mon, 17 May 2010 18:04:13 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xx.71.170:5060 --->
- SIP/2.0 200 OK
- From: "asterisk"<sip:asterisk@xx.xx.241.180>;tag=as773a250d
- To: <sip:504@192.168.1.111:5060>;tag=6f01a8c0-13c4-3c9d2-ecbd2dc-bcf
- Call-ID: 4658ee29294db8e30c6816d2609518ad@xx.xx.241.180
- CSeq: 102 OPTIONS
- Via: SIP/2.0/UDP xx.xx.241.180:5060;rport=5060;branch=z9hG4bK04978a6e
- Supported: replaces,100rel,timer
- Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS
- User-Agent: Swissvoice IP10S
- Accept: application/sdp
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Really destroying SIP dialog '4658ee29294db8e30c6816d2609518ad@xx.xx.241.180' Method: OPTIONS
- [Kmeetmecall-01*CLI>
- [May 17 20:04:16] NOTICE[3361]: pbx_spool.c:366 attempt_thread: Call completed to Local/s@gtalk_recording_local_1/n
- [Kmeetmecall-01*CLI>
- Audio is at xx.xx.241.180 port 17186
- Adding codec 0x8 (alaw) to SDP
- [Kmeetmecall-01*CLI>
- Adding non-codec 0x1 (telephone-event) to SDP
- [Kmeetmecall-01*CLI>
- Reliably Transmitting (no NAT) to xx.xxx.204.68:5060:
- INVITE sip:0621899999@xx.xxx.204.68 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK69577205;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as74578279
- To: <sip:0621899999@xx.xxx.204.68>
- Contact: <sip:asterisk@xx.xx.241.180>
- Call-ID: 356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Date: Mon, 17 May 2010 18:04:17 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Type: application/sdp
- Content-Length: 240
- v=0
- o=root 2685 2685 IN IP4 xx.xx.241.180
- s=session
- c=IN IP4 xx.xx.241.180
- t=0 0
- m=audio 17186 RTP/AVP 8 101
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.204.68:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK69577205;received=xx.xx.241.180;rport=5060
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as74578279
- To: <sip:0621899999@xx.xxx.204.68>;tag=as6a8dee5b
- Call-ID: 356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180
- CSeq: 102 INVITE
- User-Agent: itsp VoIP Interconnect
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
- Supported: replaces
- Proxy-Authenticate: Digest algorithm=MD5, realm="itsp.nl", nonce="3d0e6e4f"
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- [Kmeetmecall-01*CLI>
- Transmitting (no NAT) to xx.xxx.204.68:5060:
- ACK sip:0621899999@xx.xxx.204.68 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK69577205;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as74578279
- To: <sip:0621899999@xx.xxx.204.68>;tag=as6a8dee5b
- Contact: <sip:asterisk@xx.xx.241.180>
- Call-ID: 356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180
- CSeq: 102 ACK
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- Audio is at xx.xx.241.180 port 17186
- Adding codec 0x8 (alaw) to SDP
- [Kmeetmecall-01*CLI>
- Adding non-codec 0x1 (telephone-event) to SDP
- [Kmeetmecall-01*CLI>
- Reliably Transmitting (no NAT) to xx.xxx.204.68:5060:
- INVITE sip:0621899999@xx.xxx.204.68 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK6bea0b96;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as74578279
- To: <sip:0621899999@xx.xxx.204.68>
- Contact: <sip:asterisk@xx.xx.241.180>
- Call-ID: 356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180
- CSeq: 103 INVITE
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Proxy-Authorization: Digest username="1429999999", realm="itsp.nl", algorithm=MD5, uri="sip:0621899999@xx.xxx.204.68", nonce="3d0e6e4f", response="b244bdd1e2d5cde5a8b309f3f5bf48fc"
- Date: Mon, 17 May 2010 18:04:17 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Type: application/sdp
- Content-Length: 240
- v=0
- o=root 2685 2686 IN IP4 xx.xx.241.180
- s=session
- c=IN IP4 xx.xx.241.180
- t=0 0
- m=audio 17186 RTP/AVP 8 101
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.204.68:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK6bea0b96;received=xx.xx.241.180;rport=5060
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as74578279
- To: <sip:0621899999@xx.xxx.204.68>
- Call-ID: 356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180
- CSeq: 103 INVITE
- User-Agent: itsp VoIP Interconnect
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
- Supported: replaces
- Contact: <sip:0621899999@xx.xxx.204.68>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- [Kmeetmecall-01*CLI>
- Reliably Transmitting (no NAT) to xx.xxx.204.68:5060:
- OPTIONS sip:xx.xxx.204.68 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK47198280;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as12b9ca60
- To: <sip:xx.xxx.204.68>
- Contact: <sip:asterisk@xx.xx.241.180>
- Call-ID: 5b6e434c586970a5107b7b407656100d@xx.xx.241.180
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Date: Mon, 17 May 2010 18:04:21 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.204.68:5060 --->
- SIP/2.0 404 Not Found
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK47198280;received=xx.xx.241.180;rport=5060
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as12b9ca60
- To: <sip:xx.xxx.204.68>;tag=as64a1479f
- Call-ID: 5b6e434c586970a5107b7b407656100d@xx.xx.241.180
- CSeq: 102 OPTIONS
- User-Agent: itsp VoIP Interconnect
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
- Supported: replaces
- Accept: application/sdp
- Content-Length: 0
- [Kmeetmecall-01*CLI>
- <------------->
- --- (11 headers 0 lines) ---
- [Kmeetmecall-01*CLI>
- Really destroying SIP dialog '5b6e434c586970a5107b7b407656100d@xx.xx.241.180' Method: OPTIONS
- [Kmeetmecall-01*CLI>
- Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
- OPTIONS sip:itsp2.net:6060 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK4a1aadc7;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as0dd6cca1
- To: <sip:itsp2.net:6060>
- Contact: <sip:asterisk@xx.xx.241.180>
- Call-ID: 0497596230645a3e464687e45d8408c4@xx.xx.241.180
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Date: Mon, 17 May 2010 18:04:21 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.119.38:6060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK4a1aadc7;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as0dd6cca1
- To: <sip:itsp2.net:6060>
- Call-ID: 0497596230645a3e464687e45d8408c4@xx.xx.241.180
- CSeq: 102 OPTIONS
- Contact: <sip:xx.xxx.119.38:6060>
- User-Agent: Asterisk PBX
- Date: Mon, 17 May 2010 18:04:21 GMT
- Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
- Accept: application/sdp, application/qsig, text/plain, application/dtmf-relay
- Content-Length: 0
- <------------->
- --- (12 headers 0 lines) ---
- Really destroying SIP dialog '0497596230645a3e464687e45d8408c4@xx.xx.241.180' Method: OPTIONS
- [Kmeetmecall-01*CLI>
- Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
- OPTIONS sip:itsp2.net:6060 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK11453973;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as0d16bc6c
- To: <sip:itsp2.net:6060>
- Contact: <sip:asterisk@xx.xx.241.180>
- Call-ID: 5867f58d6c22121b684efffb4505e010@xx.xx.241.180
- CSeq: 102 OPTIONS
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Date: Mon, 17 May 2010 18:04:21 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.119.38:6060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK11453973;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as0d16bc6c
- To: <sip:itsp2.net:6060>
- Call-ID: 5867f58d6c22121b684efffb4505e010@xx.xx.241.180
- CSeq: 102 OPTIONS
- Contact: <sip:xx.xxx.119.38:6060>
- User-Agent: Asterisk PBX
- Date: Mon, 17 May 2010 18:04:21 GMT
- Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
- Accept: application/sdp, application/qsig, text/plain, application/dtmf-relay
- Content-Length: 0
- <------------->
- --- (12 headers 0 lines) ---
- Really destroying SIP dialog '5867f58d6c22121b684efffb4505e010@xx.xx.241.180' Method: OPTIONS
- [Kmeetmecall-01*CLI>
- [May 17 20:04:22] NOTICE[2708]: chan_sip.c:7753 sip_reregister: -- Re-registration for 31208080652@itsp2.net
- REGISTER 12 headers, 0 lines
- REGISTER attempt 1 to 31208080652@itsp2.net
- Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
- REGISTER sip:itsp2.net:6060 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK09bb30e0;rport
- From: <sip:31208080652@itsp2.net>;tag=as646cd458
- To: <sip:31208080652@itsp2.net>
- Call-ID: 7392a5a40317eb05014f99d419bbf5c9@xx.xx.241.180
- CSeq: 170 REGISTER
- [Kmeetmecall-01*CLI>
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Expires: 120
- Contact: <sip:s@xx.xx.241.180>
- Event: registration
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.119.38:6060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK09bb30e0;rport
- From: <sip:31208080652@itsp2.net>;tag=as646cd458
- To: <sip:31208080652@itsp2.net>
- Call-ID: 7392a5a40317eb05014f99d419bbf5c9@xx.xx.241.180
- CSeq: 170 REGISTER
- Contact: <sip:s@xx.xx.241.180>
- User-Agent: Asterisk PBX
- Expires: 60
- Event: registration
- Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
- Content-Length: 0
- <------------->
- --- (12 headers 0 lines) ---
- Scheduling destruction of SIP dialog '7392a5a40317eb05014f99d419bbf5c9@xx.xx.241.180' in 32000 ms (Method: REGISTER)
- [May 17 20:04:22] NOTICE[2708]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for itsp2.net is 60 sec (Scheduling reregistration in 45 s)
- [Kmeetmecall-01*CLI>
- [May 17 20:04:22] NOTICE[2708]: chan_sip.c:7753 sip_reregister: -- Re-registration for 31592748001@itsp2.net
- REGISTER 12 headers, 0 lines
- REGISTER attempt 1 to 31592748001@itsp2.net
- Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
- REGISTER sip:itsp2.net:6060 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK498293f1;rport
- From: <sip:31592748001@itsp2.net>;tag=as48d0e10b
- To: <sip:31592748001@itsp2.net>
- Call-ID: 2175985755b07ecd1cd804db4a708021@xx.xx.241.180
- CSeq: 170 REGISTER
- [Kmeetmecall-01*CLI>
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Expires: 120
- Contact: <sip:s@xx.xx.241.180>
- Event: registration
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.119.38:6060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK498293f1;rport
- From: <sip:31592748001@itsp2.net>;tag=as48d0e10b
- To: <sip:31592748001@itsp2.net>
- Call-ID: 2175985755b07ecd1cd804db4a708021@xx.xx.241.180
- CSeq: 170 REGISTER
- Contact: <sip:s@xx.xx.241.180>
- User-Agent: Asterisk PBX
- Expires: 60
- Event: registration
- Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
- Content-Length: 0
- <------------->
- --- (12 headers 0 lines) ---
- Scheduling destruction of SIP dialog '2175985755b07ecd1cd804db4a708021@xx.xx.241.180' in 32000 ms (Method: REGISTER)
- [May 17 20:04:22] NOTICE[2708]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for itsp2.net is 60 sec (Scheduling reregistration in 45 s)
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.204.68:5060 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK6bea0b96;received=xx.xx.241.180;rport=5060
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as74578279
- To: <sip:0621899999@xx.xxx.204.68>;tag=as25ad1b42
- Call-ID: 356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180
- CSeq: 103 INVITE
- User-Agent: itsp VoIP Interconnect
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
- Supported: replaces
- Contact: <sip:0621899999@xx.xxx.204.68>
- Content-Type: application/sdp
- Content-Length: 240
- v=0
- o=root 5032 5032 IN IP4 xx.xxx.204.68
- s=session
- c=IN IP4 xx.xxx.204.68
- t=0 0
- m=audio 14270 RTP/AVP 8 101
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------->
- --- (12 headers 12 lines) ---
- Found RTP audio format 8
- Found RTP audio format 101
- Peer audio RTP is at port xx.xxx.204.68:14270
- Found audio description format PCMA for ID 8
- Found audio description format telephone-event for ID 101
- Got unsupported a:fmtp in SDP offer
- Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- Peer audio RTP is at port xx.xxx.204.68:14270
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.204.68:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK6bea0b96;received=xx.xx.241.180;rport=5060
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as74578279
- To: <sip:0621899999@xx.xxx.204.68>;tag=as25ad1b42
- Call-ID: 356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180
- CSeq: 103 INVITE
- User-Agent: itsp VoIP Interconnect
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
- Supported: replaces
- Contact: <sip:0621899999@xx.xxx.204.68>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.204.68:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK6bea0b96;received=xx.xx.241.180;rport=5060
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as74578279
- To: <sip:0621899999@xx.xxx.204.68>;tag=as25ad1b42
- Call-ID: 356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180
- CSeq: 103 INVITE
- User-Agent: itsp VoIP Interconnect
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
- Supported: replaces
- Contact: <sip:0621899999@xx.xxx.204.68>
- C
- [Kmeetmecall-01*CLI>
- ontent-Type: application/sdp
- Content-Length: 240
- v=0
- o=root 5032 5033 IN IP4 xx.xxx.204.68
- s=session
- c=IN IP4 xx.xxx.204.68
- t=0 0
- m=audio 14270 RTP/AVP 8 101
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------->
- --- (12 headers 12 lines) ---
- Found RTP audio format 8
- Found RTP audio format 101
- Peer audio RTP is at port xx.xxx.204.68:14270
- Found audio description format PCMA for ID 8
- Found audio description format telephone-event for ID 101
- Got unsupported a:fmtp in SDP offer
- Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- Peer audio RTP is at port xx.xxx.204.68:14270
- list_route: hop: <sip:0621899999@xx.xxx.204.68>
- set_destination: Parsing <sip:0621899999@xx.xxx.204.68> for address/port to send to
- set_destination: set destination to xx.xxx.204.68, port 5060
- Transmitting (no NAT) to xx.xxx.204.68:5060:
- ACK sip:0621899999@xx.xxx.204.68 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK559d8706;rport
- From: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as74578279
- To: <sip:0621899999@xx.xxx.204.68>;tag=as25ad1b42
- Contact: <sip:asterisk@xx.xx.241.180>
- Call-ID: 356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180
- CSeq: 103 ACK
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- Really destroying SIP dialog '7f34e9117e68dad470385cab45a5e064@xx.xx.241.180' Method: REGISTER
- [Kmeetmecall-01*CLI>
- Really destroying SIP dialog '7392a5a40317eb05014f99d419bbf5c9@xx.xx.241.180' Method: REGISTER
- [Kmeetmecall-01*CLI>
- Really destroying SIP dialog '2175985755b07ecd1cd804db4a708021@xx.xx.241.180' Method: REGISTER
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.204.68:5060 --->
- BYE sip:asterisk@xx.xx.241.180 SIP/2.0
- Via: SIP/2.0/UDP xx.xxx.204.68:5060;branch=z9hG4bK08b7b869;rport
- From: <sip:0621899999@xx.xxx.204.68>;tag=as25ad1b42
- To: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as74578279
- Call-ID: 356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180
- CSeq: 102 BYE
- User-Agent: itsp VoIP Interconnect
- Max-Forwards: 70
- X-Asterisk-HangupCause: Normal Clearing
- X-Asterisk-HangupCauseCode: 16
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Sending to xx.xxx.204.68 : 5060 (no NAT)
- <--- Transmitting (no NAT) to xx.xxx.204.68:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xx.xxx.204.68:5060;branch=z9hG4bK08b7b869;received=xx.xxx.204.68;rport=5060
- From: <sip:0621899999@xx.xxx.204.68>;tag=as25ad1b42
- To: "asterisk" <sip:asterisk@xx.xx.241.180>;tag=as74578279
- Call-ID: 356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180
- CSeq: 102 BYE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------>
- [Kmeetmecall-01*CLI>
- [May 17 20:05:04] NOTICE[3375]: pbx_spool.c:366 attempt_thread: Call completed to SIP/0621899999@0852100340
- [Kmeetmecall-01*CLI>
- Really destroying SIP dialog '356ca636546daa7d7fdc582c165e4f45@xx.xx.241.180' Method: BYE
- [Kmeetmecall-01*CLI>
- [May 17 20:05:07] NOTICE[2708]: chan_sip.c:7753 sip_reregister: -- Re-registration for 31208080652@itsp2.net
- REGISTER 12 headers, 0 lines
- REGISTER attempt 1 to 31208080652@itsp2.net
- Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
- REGISTER sip:itsp2.net:6060 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK1a344c59;rport
- From: <sip:31208080652@itsp2.net>;tag=as3842065a
- To: <sip:31208080652@itsp2.net>
- Call-ID: 7392a5a40317eb05014f99d419bbf5c9@xx.xx.241.180
- CSeq: 171 REGISTER
- [Kmeetmecall-01*CLI>
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Expires: 120
- Contact: <sip:s@xx.xx.241.180>
- Event: registration
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.119.38:6060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK1a344c59;rport
- From: <sip:31208080652@itsp2.net>;tag=as3842065a
- To: <sip:31208080652@itsp2.net>
- Call-ID: 7392a5a40317eb05014f99d419bbf5c9@xx.xx.241.180
- CSeq: 171 REGISTER
- Contact: <sip:s@xx.xx.241.180>
- User-Agent: Asterisk PBX
- Expires: 60
- Event: registration
- Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
- Content-Length: 0
- <------------->
- --- (12 headers 0 lines) ---
- Scheduling destruction of SIP dialog '7392a5a40317eb05014f99d419bbf5c9@xx.xx.241.180' in 32000 ms (Method: REGISTER)
- [May 17 20:05:07] NOTICE[2708]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for itsp2.net is 60 sec (Scheduling reregistration in 45 s)
- [Kmeetmecall-01*CLI>
- [May 17 20:05:07] NOTICE[2708]: chan_sip.c:7753 sip_reregister: -- Re-registration for 31592748001@itsp2.net
- REGISTER 12 headers, 0 lines
- REGISTER attempt 1 to 31592748001@itsp2.net
- Reliably Transmitting (no NAT) to xx.xxx.119.38:6060:
- REGISTER sip:itsp2.net:6060 SIP/2.0
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK75e2c0e0;rport
- From: <sip:31592748001@itsp2.net>;tag=as777e7123
- To: <sip:31592748001@itsp2.net>
- Call-ID: 2175985755b07ecd1cd804db4a708021@xx.xx.241.180
- CSeq: 171 REGISTER
- [Kmeetmecall-01*CLI>
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Expires: 120
- Contact: <sip:s@xx.xx.241.180>
- Event: registration
- Content-Length: 0
- ---
- [Kmeetmecall-01*CLI>
- <--- SIP read from xx.xxx.119.38:6060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP xx.xx.241.180:5060;branch=z9hG4bK75e2c0e0;rport
- From: <sip:31592748001@itsp2.net>;tag=as777e7123
- To: <sip:31592748001@itsp2.net>
- Call-ID: 2175985755b07ecd1cd804db4a708021@xx.xx.241.180
- CSeq: 171 REGISTER
- Contact: <sip:s@xx.xx.241.180>
- User-Agent: Asterisk PBX
- Expires: 60
- Event: registration
- Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
- Content-Length: 0
- <------------->
- --- (12 headers 0 lines) ---
- Scheduling destruction of SIP dialog '2175985755b07ecd1cd804db4a708021@xx.xx.241.180' in 32000 ms (Method: REGISTER)
- [May 17 20:05:07] NOTICE[2708]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for itsp2.net is 60 sec (Scheduling reregistration in 45 s)
- [Kmeetmecall-01*CLI>
- Disconnected from Asterisk server
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