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- Asterisk 15.0.0, Copyright (C) 1999 - 2016, 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 15.0.0 currently running on Asty (pid = 6313)
- Asty*CLI> sip set debug on
- SIP Debugging enabled
- <--- SIP read from TCP:10.11.3.58:63611 --->
- INVITE sip:1003@110.93.203.154 SIP/2.0
- Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.dQjEOF~9I;rport
- From: <sip:God@110.93.203.154>;tag=WGTTVGU85
- To: sip:1003@110.93.203.154
- CSeq: 20 INVITE
- Call-ID: UYTfLSiokR
- Max-Forwards: 70
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Content-Type: application/sdp
- Content-Length: 954
- Contact: <sip:God@10.11.3.58:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
- User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
- v=0
- o=God 1019 1492 IN IP4 10.11.3.58
- s=Talk
- c=IN IP4 10.11.3.58
- t=0 0
- a=ice-pwd:0f022f7358cd64f6c4450f9a
- a=ice-ufrag:6947d51f
- a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
- m=audio 7078 RTP/AVPF 96 97 98 0 8 99 101 100 102 103
- c=IN IP4 110.93.203.154
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:98 speex/8000
- a=fmtp:98 vbr=on
- a=rtpmap:99 speex/32000
- a=fmtp:99 vbr=on
- a=rtpmap:101 telephone-event/48000
- a=rtpmap:100 telephone-event/16000
- a=rtpmap:102 telephone-event/8000
- a=rtpmap:103 telephone-event/32000
- a=candidate:1 1 UDP 2130706431 10.11.3.58 7078 typ host
- a=candidate:1 2 UDP 2130706430 10.11.3.58 7079 typ host
- a=candidate:2 1 UDP 1694498815 110.93.203.154 7078 typ srflx raddr 10.11.3.58 rport 7078
- a=candidate:2 2 UDP 1694498814 110.93.203.154 7079 typ srflx raddr 10.11.3.58 rport 7079
- a=rtcp-fb:* trr-int 5000
- a=rtcp-fb:* ccm tmmbr
- <------------->
- --- (13 headers 28 lines) ---
- Sending to 10.11.3.58:63611 (no NAT)
- Sending to 10.11.3.58:63611 (no NAT)
- Using INVITE request as basis request - UYTfLSiokR
- Found peer 'God' for 'God' from 10.11.3.58:63611
- <--- Reliably Transmitting (no NAT) to 10.11.3.58:63611 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.dQjEOF~9I;received=10.11.3.58;rport=63611
- From: <sip:God@110.93.203.154>;tag=WGTTVGU85
- To: sip:1003@110.93.203.154;tag=as09232fff
- Call-ID: UYTfLSiokR
- CSeq: 20 INVITE
- Server: Asterisk PBX 15.0.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59fb78f5"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'UYTfLSiokR' in 32000 ms (Method: INVITE)
- <--- SIP read from TCP:10.11.3.58:63611 --->
- ACK sip:1003@10.10.10.252:5060 SIP/2.0
- Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.dQjEOF~9I;rport
- Call-ID: UYTfLSiokR
- From: <sip:God@10.10.10.252:5060>;tag=WGTTVGU85
- To: <sip:1003@10.10.10.252:5060>;tag=as09232fff
- Contact: <sip:God@10.11.3.58:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
- Max-Forwards: 70
- CSeq: 20 ACK
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- <--- SIP read from TCP:10.11.3.58:63611 --->
- INVITE sip:1003@110.93.203.154 SIP/2.0
- Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.5fE55OBMZ;rport
- From: <sip:God@110.93.203.154>;tag=WGTTVGU85
- To: sip:1003@110.93.203.154
- CSeq: 21 INVITE
- Call-ID: UYTfLSiokR
- Max-Forwards: 70
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Content-Type: application/sdp
- Content-Length: 954
- Contact: <sip:God@10.11.3.58:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
- User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
- Authorization: Digest realm="asterisk", nonce="59fb78f5", algorithm=MD5, username="God", uri="sip:1003@110.93.203.154", response="b0de4823a7a50733150b148e2d0fb8c0"
- v=0
- o=God 1019 1492 IN IP4 10.11.3.58
- s=Talk
- c=IN IP4 10.11.3.58
- t=0 0
- a=ice-pwd:0f022f7358cd64f6c4450f9a
- a=ice-ufrag:6947d51f
- a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
- m=audio 7078 RTP/AVPF 96 97 98 0 8 99 101 100 102 103
- c=IN IP4 110.93.203.154
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:98 speex/8000
- a=fmtp:98 vbr=on
- a=rtpmap:99 speex/32000
- a=fmtp:99 vbr=on
- a=rtpmap:101 telephone-event/48000
- a=rtpmap:100 telephone-event/16000
- a=rtpmap:102 telephone-event/8000
- a=rtpmap:103 telephone-event/32000
- a=candidate:1 1 UDP 2130706431 10.11.3.58 7078 typ host
- a=candidate:1 2 UDP 2130706430 10.11.3.58 7079 typ host
- a=candidate:2 1 UDP 1694498815 110.93.203.154 7078 typ srflx raddr 10.11.3.58 rport 7078
- a=candidate:2 2 UDP 1694498814 110.93.203.154 7079 typ srflx raddr 10.11.3.58 rport 7079
- a=rtcp-fb:* trr-int 5000
- a=rtcp-fb:* ccm tmmbr
- <------------->
- --- (14 headers 28 lines) ---
- Sending to 10.11.3.58:63611 (no NAT)
- Using INVITE request as basis request - UYTfLSiokR
- Found peer 'God' for 'God' from 10.11.3.58:63611
- [Oct 31 08:08:02] NOTICE[7067][C-00011555]: chan_sip.c:10421 process_sdp: Received AVPF profile in audio offer but AVPF is not enabled, enabling: audio 7078 RTP/AVPF 96 97 98 0 8 99 101 100 102 103
- Found RTP audio format 96
- Found RTP audio format 97
- Found RTP audio format 98
- Found RTP audio format 0
- Found RTP audio format 8
- Found RTP audio format 99
- Found RTP audio format 101
- Found RTP audio format 100
- Found RTP audio format 102
- Found RTP audio format 103
- Found audio description format opus for ID 96
- Found audio description format speex for ID 97
- Found audio description format speex for ID 98
- Found audio description format speex for ID 99
- Found unknown media description format telephone-event for ID 101
- Found unknown media description format telephone-event for ID 100
- Found audio description format telephone-event for ID 102
- Found unknown media description format telephone-event for ID 103
- Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(ulaw|alaw|opus|speex16|speex|speex32)/video=(nothing)/text=(nothing), combined - (opus|speex16|speex32|ulaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 110.93.203.154:7078
- Looking for 1003 in common (domain 110.93.203.154)
- sip_route_dump: route/path hop: <sip:God@10.11.3.58:63611;transport=tcp>
- <--- Transmitting (no NAT) to 10.11.3.58:63611 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.5fE55OBMZ;received=10.11.3.58;rport=63611
- From: <sip:God@110.93.203.154>;tag=WGTTVGU85
- To: sip:1003@110.93.203.154
- Call-ID: UYTfLSiokR
- CSeq: 21 INVITE
- Server: Asterisk PBX 15.0.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:1003@10.10.10.252:5060;transport=tcp>
- Content-Length: 0
- <------------>
- Audio is at 10928
- Adding codec opus to SDP
- Adding codec speex16 to SDP
- Adding codec speex32 to SDP
- Adding codec silk24 to SDP
- Adding codec silk16 to SDP
- Adding codec ulaw to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (no NAT) to 10.11.1.100:45222:
- INVITE sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 15.0.0
- Date: Tue, 31 Oct 2017 03:08:02 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 321
- v=0
- o=root 633427397 633427397 IN IP4 10.10.10.252
- s=Asterisk PBX 15.0.0
- c=IN IP4 10.10.10.252
- t=0 0
- m=audio 10928 RTP/AVP 96 97 99 0 102
- a=rtpmap:96 opus/48000/2
- a=rtpmap:97 speex/16000
- a=rtpmap:99 speex/32000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:102 telephone-event/8000
- a=fmtp:102 0-16
- a=maxptime:60
- a=sendrecv
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- Content-Length: 0
- <------------->
- --- (7 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- sip_route_dump: no route/path
- <--- Transmitting (no NAT) to 10.11.3.58:63611 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.5fE55OBMZ;received=10.11.3.58;rport=63611
- From: <sip:God@110.93.203.154>;tag=WGTTVGU85
- To: sip:1003@110.93.203.154;tag=as5f4d7d62
- Call-ID: UYTfLSiokR
- CSeq: 21 INVITE
- Server: Asterisk PBX 15.0.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:1003@10.10.10.252:5060;transport=tcp>
- Content-Length: 0
- <------------>
- Really destroying SIP dialog 'mMhVlrBjlh' Method: REGISTER
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Content-Type: application/sdp
- Content-Length: 251
- v=0
- o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
- s=Talk
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVP 96 97 0 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:102 telephone-event/8000
- <------------->
- --- (12 headers 11 lines) ---
- Found RTP audio format 96
- Found RTP audio format 97
- Found RTP audio format 0
- Found RTP audio format 102
- Found audio description format opus for ID 96
- Found audio description format speex for ID 97
- Found audio description format telephone-event for ID 102
- Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(ulaw|opus|speex16)/video=(nothing)/text=(nothing), combined - (opus|speex16|ulaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 10.11.1.100:7076
- sip_route_dump: route/path hop: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>
- Transmitting (no NAT) to 10.11.1.100:45222:
- ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3b7a684d
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- Audio is at 11938
- Adding codec opus to SDP
- Adding codec speex16 to SDP
- Adding codec speex32 to SDP
- Adding codec ulaw to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- <--- Reliably Transmitting (no NAT) to 10.11.3.58:63611 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.5fE55OBMZ;received=10.11.3.58;rport=63611
- From: <sip:God@110.93.203.154>;tag=WGTTVGU85
- To: sip:1003@110.93.203.154;tag=as5f4d7d62
- Call-ID: UYTfLSiokR
- CSeq: 21 INVITE
- Server: Asterisk PBX 15.0.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Contact: <sip:1003@10.10.10.252:5060;transport=tcp>
- Content-Type: application/sdp
- Content-Length: 324
- v=0
- o=root 1076832751 1076832751 IN IP4 10.10.10.252
- s=Asterisk PBX 15.0.0
- c=IN IP4 10.10.10.252
- t=0 0
- m=audio 11938 RTP/AVPF 96 97 99 0 102
- a=rtpmap:96 opus/48000/2
- a=rtpmap:97 speex/16000
- a=rtpmap:99 speex/32000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:102 telephone-event/8000
- a=fmtp:102 0-16
- a=maxptime:60
- a=sendrecv
- <------------>
- Audio is at 10928
- Adding codec opus to SDP
- Adding codec speex16 to SDP
- Adding codec speex32 to SDP
- Adding codec silk24 to SDP
- Adding codec silk16 to SDP
- Adding codec ulaw to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (no NAT) to 10.11.1.100:45222:
- INVITE sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3ef7d81d
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 103 INVITE
- User-Agent: Asterisk PBX 15.0.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- X-asterisk-Info: SIP re-invite (External RTP bridge)
- Content-Type: application/sdp
- Content-Length: 324
- v=0
- o=root 633427397 633427398 IN IP4 110.93.203.154
- s=Asterisk PBX 15.0.0
- c=IN IP4 110.93.203.154
- t=0 0
- m=audio 7078 RTP/AVP 96 97 99 0 102
- a=rtpmap:96 opus/48000/2
- a=rtpmap:97 speex/16000
- a=rtpmap:99 speex/32000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:102 telephone-event/8000
- a=fmtp:102 0-16
- a=maxptime:60
- a=sendrecv
- ---
- <--- SIP read from TCP:10.11.3.58:63611 --->
- ACK sip:1003@10.10.10.252:5060;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.11.3.58:63611;rport;branch=z9hG4bK.jhhHRGJBd
- From: <sip:God@10.10.10.252:5060>;tag=WGTTVGU85
- To: <sip:1003@10.10.10.252:5060>;tag=as5f4d7d62
- CSeq: 21 ACK
- Call-ID: UYTfLSiokR
- Max-Forwards: 70
- Authorization: Digest realm="asterisk", nonce="59fb78f5", algorithm=MD5, username="God", uri="sip:1003@110.93.203.154", response="b0de4823a7a50733150b148e2d0fb8c0"
- User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3b7a684d
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3ef7d81d
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 103 INVITE
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Content-Type: application/sdp
- Content-Length: 251
- v=0
- o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
- s=Talk
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVP 96 97 0 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:102 telephone-event/8000
- <------------->
- --- (12 headers 11 lines) ---
- Transmitting (no NAT) to 10.11.1.100:45222:
- ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK4ed75a23
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK4ed75a23
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- Audio is at 11938
- Adding codec opus to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (no NAT) to 10.11.3.58:63611:
- INVITE sip:God@10.11.3.58:63611;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK4357d807;rport
- Max-Forwards: 70
- From: sip:1003@110.93.203.154;tag=as5f4d7d62
- To: <sip:God@110.93.203.154>;tag=WGTTVGU85
- Contact: <sip:1003@10.10.10.252:5060;transport=tcp>
- Call-ID: UYTfLSiokR
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 15.0.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- X-asterisk-Info: SIP re-invite (External RTP bridge)
- Content-Type: application/sdp
- Content-Length: 241
- v=0
- o=root 1076832751 1076832752 IN IP4 10.11.1.100
- s=Asterisk PBX 15.0.0
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVPF 96 102
- a=rtpmap:96 opus/48000/2
- a=rtpmap:102 telephone-event/8000
- a=fmtp:102 0-16
- a=maxptime:60
- a=sendrecv
- ---
- <--- SIP read from TCP:10.11.3.58:63611 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK4357d807;rport
- From: <sip:1003@10.10.10.252:5060>;tag=as5f4d7d62
- To: <sip:God@10.10.10.252:5060>;tag=WGTTVGU85
- Call-ID: UYTfLSiokR
- CSeq: 102 INVITE
- Content-Length: 0
- <------------->
- --- (7 headers 0 lines) ---
- <--- SIP read from TCP:10.11.3.58:63611 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK4357d807;rport
- From: <sip:1003@10.10.10.252:5060>;tag=as5f4d7d62
- To: <sip:God@10.10.10.252:5060>;tag=WGTTVGU85
- Call-ID: UYTfLSiokR
- CSeq: 102 INVITE
- User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:God@10.11.3.58:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
- Content-Type: application/sdp
- Content-Length: 219
- v=0
- o=God 1019 1494 IN IP4 10.11.3.58
- s=Talk
- c=IN IP4 10.11.3.58
- t=0 0
- m=audio 7078 RTP/AVPF 96 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:102 telephone-event/8000
- a=rtcp-fb:* trr-int 5000
- <------------->
- --- (12 headers 10 lines) ---
- Found RTP audio format 96
- Found RTP audio format 102
- Found audio description format opus for ID 96
- Found audio description format telephone-event for ID 102
- Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 10.11.3.58:7078
- Transmitting (no NAT) to 10.11.3.58:63611:
- ACK sip:God@10.11.3.58:63611;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK0f6dfe1a;rport
- Max-Forwards: 70
- From: sip:1003@110.93.203.154;tag=as5f4d7d62
- To: <sip:God@110.93.203.154>;tag=WGTTVGU85
- Contact: <sip:1003@10.10.10.252:5060;transport=tcp>
- Call-ID: UYTfLSiokR
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Content-Type: application/sdp
- Content-Length: 251
- v=0
- o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
- s=Talk
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVP 96 97 0 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:102 telephone-event/8000
- <------------->
- --- (12 headers 11 lines) ---
- Transmitting (no NAT) to 10.11.1.100:45222:
- ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK54318e30
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK54318e30
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Content-Type: application/sdp
- Content-Length: 251
- v=0
- o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
- s=Talk
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVP 96 97 0 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:102 telephone-event/8000
- <------------->
- --- (12 headers 11 lines) ---
- Transmitting (no NAT) to 10.11.1.100:45222:
- ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6cb9f866
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6cb9f866
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- REGISTER sip:10.10.10.252:5060 SIP/2.0
- Via: SIP/2.0/TCP 10.11.1.100:45222;alias;branch=z9hG4bK.GuaFIrLTw;rport
- From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=4-XZu0mXf
- To: sip:bilal.lodhia@10.10.10.252:5060
- CSeq: 24 REGISTER
- Call-ID: BcqQMUJgTI
- Max-Forwards: 70
- Supported: replaces, outbound
- Accept: application/sdp
- Accept: text/plain
- Accept: application/vnd.gsma.rcs-ft-http+xml
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Expires: 3600
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Content-Length: 0
- Authorization: Digest realm="asterisk", nonce="48627d13", algorithm=MD5, username="bilal.lodhia", uri="sip:110.93.203.154", response="0f8d439984394d69e5d1be86aacc9876"
- <------------->
- --- (16 headers 0 lines) ---
- Sending to 10.11.1.100:45222 (no NAT)
- Sending to 10.11.1.100:45222 (no NAT)
- <--- Transmitting (no NAT) to 10.11.1.100:45222 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/TCP 10.11.1.100:45222;alias;branch=z9hG4bK.GuaFIrLTw;received=10.11.1.100;rport=45222
- From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=4-XZu0mXf
- To: sip:bilal.lodhia@10.10.10.252:5060;tag=as5e6e5137
- Call-ID: BcqQMUJgTI
- CSeq: 24 REGISTER
- Server: Asterisk PBX 15.0.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="545ad159"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'BcqQMUJgTI' in 32000 ms (Method: REGISTER)
- <--- SIP read from TCP:10.11.1.100:45222 --->
- REGISTER sip:10.10.10.252:5060 SIP/2.0
- Via: SIP/2.0/TCP 10.11.1.100:45222;alias;branch=z9hG4bK.5omppCgF8;rport
- From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=4-XZu0mXf
- To: sip:bilal.lodhia@10.10.10.252:5060
- CSeq: 25 REGISTER
- Call-ID: BcqQMUJgTI
- Max-Forwards: 70
- Supported: replaces, outbound
- Accept: application/sdp
- Accept: text/plain
- Accept: application/vnd.gsma.rcs-ft-http+xml
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Expires: 3600
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Content-Length: 0
- Authorization: Digest realm="asterisk", nonce="545ad159", algorithm=MD5, username="bilal.lodhia", uri="sip:110.93.203.154", response="455196a707c686a88e86e39b62d6de77"
- <------------->
- --- (16 headers 0 lines) ---
- Sending to 10.11.1.100:45222 (no NAT)
- <--- Transmitting (no NAT) to 10.11.1.100:45222 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/TCP 10.11.1.100:45222;alias;branch=z9hG4bK.5omppCgF8;received=10.11.1.100;rport=45222
- From: <sip:bilal.lodhia@10.10.10.252:5060>;tag=4-XZu0mXf
- To: sip:bilal.lodhia@10.10.10.252:5060;tag=as5e6e5137
- Call-ID: BcqQMUJgTI
- CSeq: 25 REGISTER
- Server: Asterisk PBX 15.0.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Expires: 3600
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;expires=3600
- Date: Tue, 31 Oct 2017 03:08:12 GMT
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'BcqQMUJgTI' in 32000 ms (Method: REGISTER)
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Content-Type: application/sdp
- Content-Length: 251
- v=0
- o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
- s=Talk
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVP 96 97 0 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:102 telephone-event/8000
- <------------->
- --- (12 headers 11 lines) ---
- Transmitting (no NAT) to 10.11.1.100:45222:
- ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3489f037
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK3489f037
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Content-Type: application/sdp
- Content-Length: 251
- v=0
- o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
- s=Talk
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVP 96 97 0 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:102 telephone-event/8000
- <------------->
- --- (12 headers 11 lines) ---
- Transmitting (no NAT) to 10.11.1.100:45222:
- ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6e312c3f
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6e312c3f
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Content-Type: application/sdp
- Content-Length: 251
- v=0
- o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
- s=Talk
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVP 96 97 0 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:102 telephone-event/8000
- <------------->
- --- (12 headers 11 lines) ---
- Transmitting (no NAT) to 10.11.1.100:45222:
- ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6bab51b0
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK6bab51b0
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Content-Type: application/sdp
- Content-Length: 251
- v=0
- o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
- s=Talk
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVP 96 97 0 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:102 telephone-event/8000
- <------------->
- --- (12 headers 11 lines) ---
- Transmitting (no NAT) to 10.11.1.100:45222:
- ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK73747bf8
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK73747bf8
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Content-Type: application/sdp
- Content-Length: 251
- v=0
- o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
- s=Talk
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVP 96 97 0 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:102 telephone-event/8000
- <------------->
- --- (12 headers 11 lines) ---
- Transmitting (no NAT) to 10.11.1.100:45222:
- ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK570eb103
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK570eb103
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Content-Type: application/sdp
- Content-Length: 251
- v=0
- o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
- s=Talk
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVP 96 97 0 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:102 telephone-event/8000
- <------------->
- --- (12 headers 11 lines) ---
- Transmitting (no NAT) to 10.11.1.100:45222:
- ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK08d2b416
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK08d2b416
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5a7ece2b
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=tcp>;tag=CPj1QSM
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 INVITE
- User-Agent: LinphoneAndroid/3.3.0 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:bilal.lodhia@10.11.1.100:45222;transport=tcp>;+sip.instance="<urn:uuid:62c63056-fff5-4fb1-a974-eeac4ac343ca>"
- Content-Type: application/sdp
- Content-Length: 251
- v=0
- o=bilal.lodhia 4022 1725 IN IP4 10.11.1.100
- s=Talk
- c=IN IP4 10.11.1.100
- t=0 0
- m=audio 7076 RTP/AVP 96 97 0 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:97 speex/16000
- a=fmtp:97 vbr=on
- a=rtpmap:102 telephone-event/8000
- <------------->
- --- (12 headers 11 lines) ---
- Transmitting (no NAT) to 10.11.1.100:45222:
- ACK sip:bilal.lodhia@10.11.1.100:45222;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5bca12c4
- Max-Forwards: 70
- From: <sip:God@10.10.10.252>;tag=as11d0abee
- To: <sip:bilal.lodhia@10.11.1.100:45222;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1;transport=
- Contact: <sip:God@10.10.10.252:5060;transport=tcp>
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- SIP/2.0 400 Bad request
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK5bca12c4
- From: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 102 ACK
- Content-Length: 0
- <------------->
- --- (6 headers 0 lines) ---
- <--- SIP read from TCP:10.11.1.100:45222 --->
- BYE sip:God@10.10.10.252:5060;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.11.1.100:45222;branch=z9hG4bK.UFqQa6n~L;rport
- From: <sip:bilal.lodhia@10.11.1.100;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1>;tag=CPj1QSM
- To: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- CSeq: 111 BYE
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- Max-Forwards: 70
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- Sending to 10.11.1.100:45222 (no NAT)
- Audio is at 11938
- Adding codec opus to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (no NAT) to 10.11.3.58:63611:
- INVITE sip:God@10.11.3.58:63611;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK745f2b3a;rport
- Max-Forwards: 70
- From: sip:1003@110.93.203.154;tag=as5f4d7d62
- To: <sip:God@110.93.203.154>;tag=WGTTVGU85
- Contact: <sip:1003@10.10.10.252:5060;transport=tcp>
- Call-ID: UYTfLSiokR
- CSeq: 103 INVITE
- User-Agent: Asterisk PBX 15.0.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- X-asterisk-Info: SIP re-invite (External RTP bridge)
- Content-Type: application/sdp
- Content-Length: 244
- v=0
- o=root 1076832751 1076832753 IN IP4 10.10.10.252
- s=Asterisk PBX 15.0.0
- c=IN IP4 10.10.10.252
- t=0 0
- m=audio 11938 RTP/AVPF 96 102
- a=rtpmap:96 opus/48000/2
- a=rtpmap:102 telephone-event/8000
- a=fmtp:102 0-16
- a=maxptime:60
- a=sendrecv
- ---
- Scheduling destruction of SIP dialog 'UYTfLSiokR' in 32000 ms (Method: ACK)
- Scheduling destruction of SIP dialog '0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060' in 32000 ms (Method: BYE)
- <--- Transmitting (no NAT) to 10.11.1.100:45222 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/TCP 10.11.1.100:45222;branch=z9hG4bK.UFqQa6n~L;received=10.11.1.100;rport=45222
- From: <sip:bilal.lodhia@10.11.1.100;app-id=929724111839;pn-type=firebase;pn-tok=d1l2vUw8aik:APA91bGSW0qzVK6T3Pmd_Rx4C_CGtcXcMO-ZQA_tBF_CY5FT2bEIdLQjnuHxXeAq6YLcZy0aqMW_yThBHsFgcRHj80eh1S3bkuCEjS0f85vy3gh1RH7L5UCBVSaO8tkoEqR7dtiYyLjK;pn-silent=1>;tag=CPj1QSM
- To: <sip:God@10.10.10.252:5060>;tag=as11d0abee
- Call-ID: 0ccc376232ab0d7355d50df36960bc45@10.10.10.252:5060
- CSeq: 111 BYE
- Server: Asterisk PBX 15.0.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- <--- SIP read from TCP:10.11.3.58:63611 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK745f2b3a;rport
- From: <sip:1003@10.10.10.252:5060>;tag=as5f4d7d62
- To: <sip:God@10.10.10.252:5060>;tag=WGTTVGU85
- Call-ID: UYTfLSiokR
- CSeq: 103 INVITE
- Content-Length: 0
- <------------->
- --- (7 headers 0 lines) ---
- <--- SIP read from TCP:10.11.3.58:63611 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK745f2b3a;rport
- From: <sip:1003@10.10.10.252:5060>;tag=as5f4d7d62
- To: <sip:God@10.10.10.252:5060>;tag=WGTTVGU85
- Call-ID: UYTfLSiokR
- CSeq: 103 INVITE
- User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
- Supported: replaces, outbound
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
- Contact: <sip:God@10.11.3.58:63611;transport=tcp>;+sip.instance="<urn:uuid:a9f770a4-e6e8-4630-a581-db80ab9bcdcf>"
- Content-Type: application/sdp
- Content-Length: 219
- v=0
- o=God 1019 1496 IN IP4 10.11.3.58
- s=Talk
- c=IN IP4 10.11.3.58
- t=0 0
- m=audio 7078 RTP/AVPF 96 102
- a=rtpmap:96 opus/48000/2
- a=fmtp:96 useinbandfec=1
- a=rtpmap:102 telephone-event/8000
- a=rtcp-fb:* trr-int 5000
- <------------->
- --- (12 headers 10 lines) ---
- Found RTP audio format 96
- Found RTP audio format 102
- Found audio description format opus for ID 96
- Found audio description format telephone-event for ID 102
- Capabilities: us - (opus|speex16|speex32|silk24|silk16|ulaw), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 10.11.3.58:7078
- Transmitting (no NAT) to 10.11.3.58:63611:
- ACK sip:God@10.11.3.58:63611;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK2ce0e61c;rport
- Max-Forwards: 70
- From: sip:1003@110.93.203.154;tag=as5f4d7d62
- To: <sip:God@110.93.203.154>;tag=WGTTVGU85
- Contact: <sip:1003@10.10.10.252:5060;transport=tcp>
- Call-ID: UYTfLSiokR
- CSeq: 103 ACK
- User-Agent: Asterisk PBX 15.0.0
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.3.58:63611 --->
- BYE sip:1003@10.10.10.252:5060;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.WY1oKxTvc;rport
- From: <sip:God@10.10.10.252:5060>;tag=WGTTVGU85
- To: <sip:1003@10.10.10.252:5060>;tag=as5f4d7d62
- CSeq: 22 BYE
- Call-ID: UYTfLSiokR
- Max-Forwards: 70
- User-Agent: Linphone Desktop/4.1.1 (belle-sip/1.6.3)
- Authorization: Digest realm="asterisk", nonce="59fb78f5", algorithm=MD5, username="God", uri="sip:1003@110.93.203.154:5060;transport=tcp", response="795ced656f889150cd12c9d71901f47e"
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- Sending to 10.11.3.58:63611 (no NAT)
- Scheduling destruction of SIP dialog 'UYTfLSiokR' in 32000 ms (Method: BYE)
- <--- Transmitting (no NAT) to 10.11.3.58:63611 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/TCP 10.11.3.58:63611;branch=z9hG4bK.WY1oKxTvc;received=10.11.3.58;rport=63611
- From: <sip:God@10.10.10.252:5060>;tag=WGTTVGU85
- To: <sip:1003@10.10.10.252:5060>;tag=as5f4d7d62
- Call-ID: UYTfLSiokR
- CSeq: 22 BYE
- Server: Asterisk PBX 15.0.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- Reliably Transmitting (no NAT) to 10.11.3.58:63611:
- BYE sip:God@10.11.3.58:63611;transport=tcp SIP/2.0
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK348950c6;rport
- Max-Forwards: 70
- From: <sip:God@10.10.10.252:5060>;tag=WGTTVGU85
- To: <sip:1003@10.10.10.252:5060>;tag=as5f4d7d62
- Call-ID: UYTfLSiokR
- CSeq: 104 BYE
- User-Agent: Asterisk PBX 15.0.0
- Proxy-Authorization: Digest username="God", realm="asterisk", algorithm=MD5, uri="sip:110.93.203.154", nonce="59fb78f5", response="01c565d00b73c5085afbcff1302d214a"
- X-Asterisk-HangupCause: Normal Clearing
- X-Asterisk-HangupCauseCode: 16
- Content-Length: 0
- ---
- <--- SIP read from TCP:10.11.3.58:63611 --->
- SIP/2.0 481 Call/transaction does not exist
- Via: SIP/2.0/TCP 10.10.10.252:5060;branch=z9hG4bK348950c6;rport
- From: <sip:God@10.10.10.252:5060>;tag=WGTTVGU85
- To: <sip:1003@10.10.10.252:5060>;tag=as5f4d7d62
- Call-ID: UYTfLSiokR
- CSeq: 104 BYE
- Content-Length: 0
- <------------->
- --- (7 headers 0 lines) ---
- Really destroying SIP dialog 'BcqQMUJgTI' Method: REGISTER
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