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One way voice SIP SDP body keep showing internal IP inste... ยป asterisk CLI SIP Debug.txt

SIP Debug Log - Edward T, 06/02/2021 10:32 PM

 
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-- Called 99995474470533@default
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    -- Executing [99995474470533@default:1] Dial("Local/99995474470533@default-00000006;2", "SIP/5474470533@kamailio,,tTo") in new stack
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  == Using SIP RTP CoS mark 5
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Audio is at 47536
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Adding codec opus to SDP
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Adding codec ulaw to SDP
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Adding non-codec 0x1 (telephone-event) to SDP
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Reliably Transmitting (NAT) to 127.0.0.1:5060:
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INVITE sip:5474470533@127.0.0.1 SIP/2.0
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Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK7e99ce53;rport
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Max-Forwards: 70
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From: "S2106022200068600051" <sip:0340518900@127.0.0.1:5070>;tag=as5ec3ee2e
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To: <sip:5474470533@127.0.0.1>
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Contact: <sip:0340518900@127.0.0.1:5070>
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Call-ID: 4eb142d9604a97676d64a372199769dc@127.0.0.1:5070
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CSeq: 102 INVITE
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User-Agent: Asterisk PBX 13.17.2-vici
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Date: Wed, 02 Jun 2021 14:00:06 GMT
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Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
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Supported: replaces
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Remote-Party-ID: "S2106022200068600051" <sip:0340518900@127.0.0.1>;party=calling;privacy=off;screen=no
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Content-Type: application/sdp
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Content-Length: 306
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v=0
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o=root 210085128 210085128 IN IP4 127.0.0.1
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s=Asterisk PBX 13.17.2-vici
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c=IN IP4 127.0.0.1
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t=0 0
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m=audio 47536 RTP/AVP 107 0 101
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a=rtpmap:107 opus/48000/2
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a=fmtp:107 useinbandfec=1
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a=rtpmap:0 PCMU/8000
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a=rtpmap:101 telephone-event/8000
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a=fmtp:101 0-16
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a=ptime:20
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a=maxptime:20
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a=sendrecv
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---
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    -- Called SIP/5474470533@kamailio
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<--- SIP read from UDP:127.0.0.1:5060 --->
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SIP/2.0 100 trying -- your call is important to us
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Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK7e99ce53;rport=5070;received=127.0.0.1
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From: "S2106022200068600051" <sip:0340518900@127.0.0.1:5070>;tag=as5ec3ee2e
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To: <sip:5474470533@127.0.0.1>
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Call-ID: 4eb142d9604a97676d64a372199769dc@127.0.0.1:5070
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CSeq: 102 INVITE
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Content-Length: 0
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<------------->
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--- (7 headers 0 lines) ---
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<--- SIP read from UDP:127.0.0.1:5060 --->
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SIP/2.0 180 Ringing
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Via: SIP/2.0/UDP 127.0.0.1:5070;rport=5070;received=127.0.0.1;branch=z9hG4bK7e99ce53
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Record-Route: <sip:211.25.xx.xx:5060;lr;r2=on>
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Record-Route: <sip:127.0.0.1;lr;r2=on>
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Call-ID: 4eb142d9604a97676d64a372199769dc@127.0.0.1:5070
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From: "S2106022200068600051" <sip:0340518900@127.0.0.1>;tag=as5ec3ee2e
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To: <sip:5474470533@127.0.0.1>;tag=KJ6m1dAMFLw63qd6Nx30ohXhaDDbxG3X
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CSeq: 102 INVITE
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Contact: "5474470533" <sip:5474470533@115.132.133.184:59009;ob>
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Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
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Content-Length: 0
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<------------->
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--- (11 headers 0 lines) ---
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sip_route_dump: route/path hop: <sip:127.0.0.1;lr;r2=on>
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sip_route_dump: route/path hop: <sip:211.25.xx.xx:5060;lr;r2=on>
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    -- SIP/kamailio-00000005 is ringing
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    -- Local/99995474470533@default-00000006;1 is ringing
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  == Manager 'sendcron' logged on from 127.0.0.1
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  == Manager 'sendcron' logged off from 127.0.0.1
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<--- SIP read from UDP:127.0.0.1:5060 --->
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SIP/2.0 200 OK
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Via: SIP/2.0/UDP 127.0.0.1:5070;rport=5070;received=127.0.0.1;branch=z9hG4bK7e99ce53
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Record-Route: <sip:211.25.xx.xx:5060;lr;r2=on>
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Record-Route: <sip:127.0.0.1;lr;r2=on>
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Call-ID: 4eb142d9604a97676d64a372199769dc@127.0.0.1:5070
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From: "S2106022200068600051" <sip:0340518900@127.0.0.1>;tag=as5ec3ee2e
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To: <sip:5474470533@127.0.0.1>;tag=KJ6m1dAMFLw63qd6Nx30ohXhaDDbxG3X
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CSeq: 102 INVITE
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Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
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Contact: "5474470533" <sip:5474470533@115.132.133.184:59009;ob>
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Supported: replaces, 100rel, norefersub
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Content-Type: application/sdp
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Content-Length: 491
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Session-Expires: 90;refresher=uac
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v=0
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o=- 3831631206 3831631207 IN IP4 10.100.1.11
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s=pjmedia
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b=AS:117
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t=0 0
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a=X-nat:0
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m=audio 30070 RTP/AVP 107 101
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c=IN IP4 10.100.1.11
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b=TIAS:96000
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a=ssrc:1930347664 cname:6995a62a5fd0458d
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a=rtpmap:107 opus/48000/2
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a=rtpmap:101 telephone-event/8000
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a=fmtp:107 useinbandfec=1
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a=fmtp:101 0-16
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a=sendrecv
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a=rtcp:30071
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a=ptime:20
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a=candidate:hVGvyh9hGu8X5XwX 1 UDP 2130706431 10.100.1.11 30070 typ host
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a=candidate:hVGvyh9hGu8X5XwX 2 UDP 2130706430 10.100.1.11 30071 typ host
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<------------->
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--- (14 headers 19 lines) ---
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Found RTP audio format 107
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Found RTP audio format 101
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Found audio description format opus for ID 107
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Found audio description format telephone-event for ID 101
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Capabilities: us - (opus|ulaw), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus)
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Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
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       > 0x18d76f0 -- Strict RTP learning after remote address set to: 10.100.1.11:30070
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Peer audio RTP is at port 10.100.1.11:30070
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sip_route_dump: route/path hop: <sip:127.0.0.1;lr;r2=on>
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sip_route_dump: route/path hop: <sip:211.25.xx.xx:5060;lr;r2=on>
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Transmitting (NAT) to 127.0.0.1:5060:
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ACK sip:5474470533@115.132.133.184:59009;ob SIP/2.0
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Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK396c894d;rport
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Route: <sip:127.0.0.1;lr;r2=on>,<sip:211.25.xx.xx:5060;lr;r2=on>
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Max-Forwards: 70
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From: "S2106022200068600051" <sip:0340518900@127.0.0.1:5070>;tag=as5ec3ee2e
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To: <sip:5474470533@127.0.0.1>;tag=KJ6m1dAMFLw63qd6Nx30ohXhaDDbxG3X
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Contact: <sip:0340518900@127.0.0.1:5070>
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Call-ID: 4eb142d9604a97676d64a372199769dc@127.0.0.1:5070
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CSeq: 102 ACK
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User-Agent: Asterisk PBX 13.17.2-vici
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Content-Length: 0
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