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- <--- SIP read from UDP:192.168.1.138:5060 --->
- INVITE sip:*2*220000@192.168.1.248 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.138:5060;rport;branch=z9hG4bKPj97dfe1bf-424f-4d92-9337-513e4b1f9160
- Max-Forwards: 70
- From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
- To: <sip:*2*220000@192.168.1.248>
- Contact: <sip:805@192.168.1.138:5060>
- Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
- CSeq: 31252 INVITE
- Subject: Phone call
- Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
- Supported: replaces, 100rel
- Content-Type: application/sdp
- Content-Length: 405
- v=0
- o=master-pc 3614303752 0 IN IP4 192.168.1.138
- s=sflphone
- c=IN IP4 192.168.1.138
- t=0 0
- m=audio 27250 RTP/AVP 0 3 8 9 110 111 112 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:9 G722/8000
- a=rtpmap:110 speex/8000
- a=rtpmap:111 speex/16000
- a=rtpmap:112 speex/32000
- a=sendrecv
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=rtcp:27251 IN IP4 192.168.1.138
- <------------->
- --- (13 headers 17 lines) ---
- Sending to 192.168.1.138:5060 (no NAT)
- Sending to 192.168.1.138:5060 (no NAT)
- Using INVITE request as basis request - e1acda66-823b-46f6-9354-4d8bdf3638b0
- Found peer '805' for '805' from 192.168.1.138:5060
- <--- Reliably Transmitting (no NAT) to 192.168.1.138:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 192.168.1.138:5060;branch=z9hG4bKPj97dfe1bf-424f-4d92-9337-513e4b1f9160;received=192.168.1.138;rport=5060
- From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
- To: <sip:*2*220000@192.168.1.248>;tag=as7568ea5f
- Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
- CSeq: 31252 INVITE
- Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54556ff6"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'e1acda66-823b-46f6-9354-4d8bdf3638b0' in 6400 ms (Method: INVITE)
- <--- SIP read from UDP:192.168.1.138:5060 --->
- ACK sip:*2*220000@192.168.1.248 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.138:5060;rport;branch=z9hG4bKPj97dfe1bf-424f-4d92-9337-513e4b1f9160
- Max-Forwards: 70
- From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
- To: <sip:*2*220000@192.168.1.248>;tag=as7568ea5f
- Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
- CSeq: 31252 ACK
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- <--- SIP read from UDP:192.168.1.138:5060 --->
- INVITE sip:*2*220000@192.168.1.248 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.138:5060;rport;branch=z9hG4bKPj4dc7b015-63f8-45ff-b82d-60bb906064d3
- Max-Forwards: 70
- From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
- To: <sip:*2*220000@192.168.1.248>
- Contact: <sip:805@192.168.1.138:5060>
- Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
- CSeq: 31253 INVITE
- Subject: Phone call
- Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
- Supported: replaces, 100rel
- Authorization: Digest username="805", realm="asterisk", nonce="54556ff6", uri="sip:*2*220000@192.168.1.248", response="390c7717e5786714118d5bcc22096a34", algorithm=MD5
- Content-Type: application/sdp
- Content-Length: 405
- v=0
- o=master-pc 3614303752 0 IN IP4 192.168.1.138
- s=sflphone
- c=IN IP4 192.168.1.138
- t=0 0
- m=audio 27250 RTP/AVP 0 3 8 9 110 111 112 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:9 G722/8000
- a=rtpmap:110 speex/8000
- a=rtpmap:111 speex/16000
- a=rtpmap:112 speex/32000
- a=sendrecv
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=rtcp:27251 IN IP4 192.168.1.138
- <------------->
- --- (14 headers 17 lines) ---
- Sending to 192.168.1.138:5060 (no NAT)
- Using INVITE request as basis request - e1acda66-823b-46f6-9354-4d8bdf3638b0
- Found peer '805' for '805' from 192.168.1.138:5060
- == Using SIP RTP CoS mark 5
- Found RTP audio format 0
- Found RTP audio format 3
- Found RTP audio format 8
- Found RTP audio format 9
- Found RTP audio format 110
- Found RTP audio format 111
- Found RTP audio format 112
- Found RTP audio format 101
- Found audio description format PCMU for ID 0
- Found audio description format GSM for ID 3
- Found audio description format PCMA for ID 8
- Found audio description format G722 for ID 9
- Found audio description format speex for ID 110
- Found audio description format speex for ID 111
- Found audio description format speex for ID 112
- Found audio description format telephone-event for ID 101
- Capabilities: us - (alaw), peer - audio=(gsm|ulaw|alaw|speex|speex16|g722|speex32)/video=(nothing)/text=(nothing), combined - (alaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 192.168.1.138:27250
- Looking for *2*220000 in office-phones (domain 192.168.1.248)
- list_route: hop: <sip:805@192.168.1.138:5060>
- <--- Transmitting (no NAT) to 192.168.1.138:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.1.138:5060;branch=z9hG4bKPj4dc7b015-63f8-45ff-b82d-60bb906064d3;received=192.168.1.138;rport=5060
- From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
- To: <sip:*2*220000@192.168.1.248>
- Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
- CSeq: 31253 INVITE
- Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:*2*220000@192.168.1.248:5060>
- Content-Length: 0
- <------------>
- -- Executing [*2*220000@office-phones:1] NoOp("SIP/805-00000002", "") in new stack
- -- Executing [*2*220000@office-phones:2] Dial("SIP/805-00000002", "SIP/teledyne/220000") in new stack
- == Using SIP RTP CoS mark 5
- Audio is at 12140
- Adding codec 100004 (alaw) to SDP
- Adding codec 100002 (gsm) to SDP
- Adding codec 100003 (ulaw) to SDP
- Adding codec 100017 (testlaw) to SDP
- Reliably Transmitting (no NAT) to 91.142.144.132:5060:
- INVITE sip:220000@91.142.144.132 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.248:5060;branch=z9hG4bK335c0318
- Max-Forwards: 70
- From: <sip:NAME@tsl.ru>;tag=as226b3fd8
- To: <sip:220000@91.142.144.132>
- Contact: <sip:NAME@192.168.1.248:5060>
- Call-ID: 6ac50c8e0e12cc4506a79f7e1e1020bb@tsl.ru
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
- Date: Mon, 14 Jul 2014 05:15:42 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 240
- v=0
- o=root 769084428 769084428 IN IP4 192.168.1.248
- s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
- c=IN IP4 192.168.1.248
- t=0 0
- m=audio 12140 RTP/AVP 8 3 0
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:0 PCMU/8000
- a=ptime:20
- a=sendrecv
- ---
- -- Called SIP/teledyne/220000
- <--- SIP read from UDP:91.142.144.132:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.1.248:5060;received=195.208.35.31;branch=z9hG4bK335c0318;rport=5060
- From: <sip:NAME@tsl.ru>;tag=as226b3fd8
- To: <sip:220000@91.142.144.132>
- Call-ID: 6ac50c8e0e12cc4506a79f7e1e1020bb@tsl.ru
- CSeq: 102 INVITE
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from UDP:91.142.144.132:5060 --->
- SIP/2.0 604 Does not exist anywhere
- Via: SIP/2.0/UDP 192.168.1.248:5060;received=195.208.35.31;branch=z9hG4bK335c0318;rport=5060
- From: <sip:NAME@tsl.ru>;tag=as226b3fd8
- To: <sip:220000@91.142.144.132>;tag=1877391753-1405314942990
- Call-ID: 6ac50c8e0e12cc4506a79f7e1e1020bb@tsl.ru
- CSeq: 102 INVITE
- Content-Length: 0
- <------------->
- --- (7 headers 0 lines) ---
- -- Got SIP response 604 "Does not exist anywhere" back from 91.142.144.132:5060
- Transmitting (no NAT) to 91.142.144.132:5060:
- ACK sip:220000@91.142.144.132 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.248:5060;branch=z9hG4bK335c0318
- Max-Forwards: 70
- From: <sip:NAME@tsl.ru>;tag=as226b3fd8
- To: <sip:220000@91.142.144.132>;tag=1877391753-1405314942990
- Contact: <sip:NAME@192.168.1.248:5060>
- Call-ID: 6ac50c8e0e12cc4506a79f7e1e1020bb@tsl.ru
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
- Content-Length: 0
- ---
- == Everyone is busy/congested at this time (1:0/0/1)
- -- Auto fallthrough, channel 'SIP/805-00000002' status is 'CHANUNAVAIL'
- <--- Reliably Transmitting (no NAT) to 192.168.1.138:5060 --->
- SIP/2.0 503 Service Unavailable
- Via: SIP/2.0/UDP 192.168.1.138:5060;branch=z9hG4bKPj4dc7b015-63f8-45ff-b82d-60bb906064d3;received=192.168.1.138;rport=5060
- From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
- To: <sip:*2*220000@192.168.1.248>;tag=as3d3d78c5
- Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
- CSeq: 31253 INVITE
- Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- X-Asterisk-HangupCause: Unallocated (unassigned) number
- X-Asterisk-HangupCauseCode: 1
- Content-Length: 0
- <------------>
- <--- SIP read from UDP:192.168.1.138:5060 --->
- ACK sip:*2*220000@192.168.1.248 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.138:5060;rport;branch=z9hG4bKPj4dc7b015-63f8-45ff-b82d-60bb906064d3
- Max-Forwards: 70
- From: <sip:805@192.168.1.248>;tag=ce98cd3c-2ec9-4020-ad36-8f7934ad353e
- To: <sip:*2*220000@192.168.1.248>;tag=as3d3d78c5
- Call-ID: e1acda66-823b-46f6-9354-4d8bdf3638b0
- CSeq: 31253 ACK
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- Really destroying SIP dialog '6ac50c8e0e12cc4506a79f7e1e1020bb@tsl.ru' Method: INVITE
- Really destroying SIP dialog 'e1acda66-823b-46f6-9354-4d8bdf3638b0' Method: ACK
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