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1
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-- Called 99995474470533@default
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2
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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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3
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== Using SIP RTP CoS mark 5
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4
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Audio is at 47536
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5
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Adding codec opus to SDP
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6
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Adding codec ulaw to SDP
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7
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Adding non-codec 0x1 (telephone-event) to SDP
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8
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Reliably Transmitting (NAT) to 127.0.0.1:5060:
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9
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INVITE sip:[email protected] SIP/2.0
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10
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Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK7e99ce53;rport
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11
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Max-Forwards: 70
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12
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From: "S2106022200068600051" <sip:[email protected]:5070>;tag=as5ec3ee2e
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13
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To: <sip:[email protected]>
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14
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Contact: <sip:[email protected]:5070>
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15
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Call-ID: [email protected]:5070
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16
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CSeq: 102 INVITE
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17
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User-Agent: Asterisk PBX 13.17.2-vici
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18
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Date: Wed, 02 Jun 2021 14:00:06 GMT
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19
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Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
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20
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Supported: replaces
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21
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Remote-Party-ID: "S2106022200068600051" <sip:[email protected]>;party=calling;privacy=off;screen=no
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22
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Content-Type: application/sdp
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23
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Content-Length: 306
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24
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25
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v=0
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26
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o=root 210085128 210085128 IN IP4 127.0.0.1
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27
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s=Asterisk PBX 13.17.2-vici
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28
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c=IN IP4 127.0.0.1
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29
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t=0 0
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30
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m=audio 47536 RTP/AVP 107 0 101
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31
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a=rtpmap:107 opus/48000/2
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32
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a=fmtp:107 useinbandfec=1
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33
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a=rtpmap:0 PCMU/8000
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34
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a=rtpmap:101 telephone-event/8000
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35
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a=fmtp:101 0-16
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36
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a=ptime:20
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37
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a=maxptime:20
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38
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a=sendrecv
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39
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|
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40
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---
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41
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-- Called SIP/5474470533@kamailio
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42
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|
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43
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<--- SIP read from UDP:127.0.0.1:5060 --->
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44
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SIP/2.0 100 trying -- your call is important to us
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45
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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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46
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From: "S2106022200068600051" <sip:[email protected]:5070>;tag=as5ec3ee2e
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47
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To: <sip:[email protected]>
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48
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Call-ID: [email protected]:5070
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49
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CSeq: 102 INVITE
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50
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Content-Length: 0
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51
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|
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52
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<------------->
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53
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--- (7 headers 0 lines) ---
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54
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|
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55
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<--- SIP read from UDP:127.0.0.1:5060 --->
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56
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SIP/2.0 180 Ringing
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57
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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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58
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Record-Route: <sip:211.25.xx.xx:5060;lr;r2=on>
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59
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Record-Route: <sip:127.0.0.1;lr;r2=on>
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60
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Call-ID: [email protected]:5070
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61
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From: "S2106022200068600051" <sip:[email protected]>;tag=as5ec3ee2e
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62
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To: <sip:[email protected]>;tag=KJ6m1dAMFLw63qd6Nx30ohXhaDDbxG3X
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63
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CSeq: 102 INVITE
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|
64
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Contact: "5474470533" <sip:[email protected]:59009;ob>
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65
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Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
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66
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Content-Length: 0
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67
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|
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68
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<------------->
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69
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--- (11 headers 0 lines) ---
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70
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sip_route_dump: route/path hop: <sip:127.0.0.1;lr;r2=on>
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71
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sip_route_dump: route/path hop: <sip:211.25.xx.xx:5060;lr;r2=on>
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72
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-- SIP/kamailio-00000005 is ringing
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73
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-- Local/99995474470533@default-00000006;1 is ringing
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74
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== Manager 'sendcron' logged on from 127.0.0.1
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75
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== Manager 'sendcron' logged off from 127.0.0.1
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76
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|
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77
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<--- SIP read from UDP:127.0.0.1:5060 --->
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78
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SIP/2.0 200 OK
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79
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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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80
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Record-Route: <sip:211.25.xx.xx:5060;lr;r2=on>
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81
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Record-Route: <sip:127.0.0.1;lr;r2=on>
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82
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Call-ID: [email protected]:5070
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83
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From: "S2106022200068600051" <sip:[email protected]>;tag=as5ec3ee2e
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|
84
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To: <sip:[email protected]>;tag=KJ6m1dAMFLw63qd6Nx30ohXhaDDbxG3X
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|
85
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CSeq: 102 INVITE
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86
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Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
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87
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Contact: "5474470533" <sip:[email protected]:59009;ob>
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88
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Supported: replaces, 100rel, norefersub
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89
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Content-Type: application/sdp
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|
90
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Content-Length: 491
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|
91
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Session-Expires: 90;refresher=uac
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|
92
|
|
|
93
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v=0
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94
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o=- 3831631206 3831631207 IN IP4 10.100.1.11
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|
95
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s=pjmedia
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96
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b=AS:117
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97
|
t=0 0
|
|
98
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a=X-nat:0
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|
99
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m=audio 30070 RTP/AVP 107 101
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|
100
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c=IN IP4 10.100.1.11
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101
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b=TIAS:96000
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102
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a=ssrc:1930347664 cname:6995a62a5fd0458d
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|
103
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a=rtpmap:107 opus/48000/2
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|
104
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a=rtpmap:101 telephone-event/8000
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|
105
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a=fmtp:107 useinbandfec=1
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|
106
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a=fmtp:101 0-16
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|
107
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a=sendrecv
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|
108
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a=rtcp:30071
|
|
109
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a=ptime:20
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|
110
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a=candidate:hVGvyh9hGu8X5XwX 1 UDP 2130706431 10.100.1.11 30070 typ host
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|
111
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a=candidate:hVGvyh9hGu8X5XwX 2 UDP 2130706430 10.100.1.11 30071 typ host
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|
112
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<------------->
|
|
113
|
--- (14 headers 19 lines) ---
|
|
114
|
Found RTP audio format 107
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|
115
|
Found RTP audio format 101
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116
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Found audio description format opus for ID 107
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117
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Found audio description format telephone-event for ID 101
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|
118
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Capabilities: us - (opus|ulaw), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus)
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|
119
|
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
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|
120
|
> 0x18d76f0 -- Strict RTP learning after remote address set to: 10.100.1.11:30070
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|
121
|
Peer audio RTP is at port 10.100.1.11:30070
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|
122
|
sip_route_dump: route/path hop: <sip:127.0.0.1;lr;r2=on>
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|
123
|
sip_route_dump: route/path hop: <sip:211.25.xx.xx:5060;lr;r2=on>
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|
124
|
Transmitting (NAT) to 127.0.0.1:5060:
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125
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ACK sip:[email protected]:59009;ob SIP/2.0
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126
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Via: SIP/2.0/UDP 127.0.0.1:5070;branch=z9hG4bK396c894d;rport
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|
127
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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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|
128
|
Max-Forwards: 70
|
|
129
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From: "S2106022200068600051" <sip:[email protected]:5070>;tag=as5ec3ee2e
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|
130
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To: <sip:[email protected]>;tag=KJ6m1dAMFLw63qd6Nx30ohXhaDDbxG3X
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|
131
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Contact: <sip:[email protected]:5070>
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|
132
|
Call-ID: [email protected]:5070
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|
133
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CSeq: 102 ACK
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|
134
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User-Agent: Asterisk PBX 13.17.2-vici
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135
|
Content-Length: 0
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