Advertisement
Not a member of Pastebin yet?
Sign Up,
it unlocks many cool features!
- <--- SIP read from UDP:78.23.152.207:36506 --->
- <------------->
- <--- SIP read from UDP:208.51.64.179:5060 --->
- INVITE sip:s@178.32.248.208:5060 SIP/2.0
- From:+32496711491 <sip:+32496711491@sonetel.net>;tag=17483246313536411F076A01
- To:vincent.iserentant <sip:vincent.iserentant@gmail.com>
- Request-Disposition:no-fork
- Expires:3600
- Content-Type:application/sdp
- Contact:sip:SonetelPBX@208.51.64.179:5060
- Call-ID:0204F756E091400000000726@F05
- Max-Forwards:69
- CSeq:1 INVITE
- Route:<sip:178.32.248.208:5060;lr>
- Via:SIP/2.0/UDP 208.51.64.179:5060;branch=z9hG4bK793425FF0169AC3E545675B024BDB879;rport
- Via:SIP/2.0/UDP 192.168.6.25:5064;branch=z9hG4bK3A8EFC19D031E7505BEF296ED5925090;received=192.168.6.25
- Content-Length:321
- Record-Route:<sip:208.51.64.179:5060;lr>
- v=0
- o=root 928832435 928832435 IN IP4 81.201.82.169
- s=session
- c=IN IP4 81.201.82.169
- t=0 0
- m=audio 10830 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=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------->
- --- (15 headers 15 lines) ---
- Sending to 208.51.64.179:5060 (NAT)
- Using INVITE request as basis request - 0204F756E091400000000726@F05
- No matching peer for '+32496711491' from '208.51.64.179:5060'
- == Using SIP RTP TOS bits 184
- == Using SIP RTP CoS mark 5
- Found RTP audio format 8
- Found RTP audio format 0
- Found RTP audio format 18
- Found RTP audio format 101
- Found audio description format PCMA for ID 8
- Found audio description format PCMU for ID 0
- Found audio description format G729 for ID 18
- Found audio description format telephone-event for ID 101
- Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
- on-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 81.201.82.169:10830
- Looking for s in from-sip-external (domain 178.32.248.208)
- list_route: hop: <sip:208.51.64.179:5060;lr>
- <--- Transmitting (NAT) to 208.51.64.179:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 208.51.64.179:5060;branch=z9hG4bK793425FF0169AC3E545675B024BDB879;received=208.51.64.179;rport=5060
- Via: SIP/2.0/UDP 192.168.6.25:5064;branch=z9hG4bK3A8EFC19D031E7505BEF296ED5925090;received=192.168.6.25
- Record-Route: <sip:208.51.64.179:5060;lr>
- From: +32496711491 <sip:+32496711491@sonetel.net>;tag=17483246313536411F076A01
- To: vincent.iserentant <sip:vincent.iserentant@gmail.com>
- Call-ID: 0204F756E091400000000726@F05
- CSeq: 1 INVITE
- Server: FPBX-2.11.0(1.8.10.1)
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:s@178.32.248.208:5060>
- Content-Length: 0
- <------------>
- -- Executing [s@from-sip-external:1] GotoIf("SIP/sonetel.net-00002855", "0?checklang:noanonymous") in new stack
- -- Goto (from-sip-external,s,5)
- -- Executing [s@from-sip-external:5] Set("SIP/sonetel.net-00002855", "TIMEOUT(absolute)=15") in new stack
- Channel will hangup at 2014-01-21 20:30:27.507 CET.
- -- Executing [s@from-sip-external:6] Log("SIP/sonetel.net-00002855", "WARNING,"Rejecting unknown SIP connection from 208.51.64.179"") in new stack
- [2014-01-21 20:30:12] WARNING[20864]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from 208.51.64.179"
- -- Executing [s@from-sip-external:7] Answer("SIP/sonetel.net-00002855", "") in new stack
- Audio is at 11320
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x8 (alaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- <--- Reliably Transmitting (NAT) to 208.51.64.179:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 208.51.64.179:5060;branch=z9hG4bK793425FF0169AC3E545675B024BDB879;received=208.51.64.179;rport=5060
- Via: SIP/2.0/UDP 192.168.6.25:5064;branch=z9hG4bK3A8EFC19D031E7505BEF296ED5925090;received=192.168.6.25
- Record-Route: <sip:208.51.64.179:5060;lr>
- From: +32496711491 <sip:+32496711491@sonetel.net>;tag=17483246313536411F076A01
- To: vincent.iserentant <sip:vincent.iserentant@gmail.com>;tag=as49ef19cb
- Call-ID: 0204F756E091400000000726@F05
- CSeq: 1 INVITE
- Server: FPBX-2.11.0(1.8.10.1)
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:s@178.32.248.208:5060>
- Content-Type: application/sdp
- Content-Length: 279
- v=0
- o=root 2062968190 2062968190 IN IP4 178.32.248.208
- s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
- c=IN IP4 178.32.248.208
- t=0 0
- m=audio 11320 RTP/AVP 0 8 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------>
- <--- SIP read from UDP:208.51.64.179:5060 --->
- ACK sip:vincent.iserentant@178.32.248.208:5060 SIP/2.0
- Contact:sip:SonetelPBX@208.51.64.179:5060
- CSeq:1 ACK
- To:vincent.iserentant <sip:vincent.iserentant@gmail.com>;tag=as49ef19cb
- From:+32496711491 <sip:+32496711491@sonetel.net>;tag=17483246313536411F076A01
- Call-ID:0204F756E091400000000726@F05
- Max-Forwards:69
- Via:SIP/2.0/UDP 208.51.64.179:5060;branch=z9hG4bKB4FA5E40E7081555D71C3E70098A3165;rport
- Via:SIP/2.0/UDP 192.168.6.25:5064;branch=z9hG4bKC952125A83A12DCD7FBF1DBBE710892C;received=192.168.6.25
- Content-Length:0
- Route:<sip:178.32.248.208:5060;lr>;server=LS
- <------------->
- --- (11 headers 0 lines) ---
- -- Executing [s@from-sip-external:8] Wait("SIP/sonetel.net-00002855", "2") in new stack
- [2014-01-21 20:30:12] NOTICE[20864]: channel.c:4149 __ast_read: Dropping incompatible voice frame on SIP/sonetel.net-00002855 of format ulaw since our native format has changed to 0x8 (alaw)
- Really destroying SIP dialog '1da05ae7-5c51f17-52def4d7@178.32.248.208' Method: REGISTER
- -- Executing [s@from-sip-external:9] Playback("SIP/sonetel.net-00002855", "ss-noservice") in new stack
- -- <SIP/sonetel.net-00002855> Playing 'ss-noservice.gsm' (language 'en')
- <--- SIP read from UDP:208.51.64.179:5060 --->
- BYE sip:vincent.iserentant@178.32.248.208:5060 SIP/2.0
- Contact:sip:SonetelPBX@208.51.64.179:5060
- To:vincent.iserentant <sip:vincent.iserentant@gmail.com>;tag=as49ef19cb
- From:+32496711491 <sip:+32496711491@sonetel.net>;tag=17483246313536411F076A01
- Call-ID:0204F756E091400000000726@F05
- Max-Forwards:69
- CSeq:2 BYE
- Via:SIP/2.0/UDP 208.51.64.179:5060;branch=z9hG4bKF8B16F68E521F8977F50B41DDC6F6DBB;rport
- Via:SIP/2.0/UDP 192.168.6.25:5064;branch=z9hG4bKB45971CA1AF74FC5A2F4AD660DEEFE2F;received=192.168.6.25
- Content-Length:0
- Route:<sip:178.32.248.208:5060;lr>;server=LS
- Record-Route:<sip:208.51.64.179:5060;lr>
- <------------->
- --- (12 headers 0 lines) ---
- Sending to 208.51.64.179:5060 (NAT)
- Scheduling destruction of SIP dialog '0204F756E091400000000726@F05' in 32000 ms (Method: BYE)
- <--- Transmitting (NAT) to 208.51.64.179:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 208.51.64.179:5060;branch=z9hG4bKF8B16F68E521F8977F50B41DDC6F6DBB;received=208.51.64.179;rport=5060
- Via: SIP/2.0/UDP 192.168.6.25:5064;branch=z9hG4bKB45971CA1AF74FC5A2F4AD660DEEFE2F;received=192.168.6.25
- Record-Route: <sip:208.51.64.179:5060;lr>
- From: +32496711491 <sip:+32496711491@sonetel.net>;tag=17483246313536411F076A01
- To: vincent.iserentant <sip:vincent.iserentant@gmail.com>;tag=as49ef19cb
- Call-ID: 0204F756E091400000000726@F05
- CSeq: 2 BYE
- Server: FPBX-2.11.0(1.8.10.1)
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- == Spawn extension (from-sip-external, s, 9) exited non-zero on 'SIP/sonetel.net-00002855'
- -- Executing [h@from-sip-external:1] Hangup("SIP/sonetel.net-00002855", "") in new stack
- == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/sonetel.net-00002855'
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement