Created
February 25, 2018 01:01
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Linphone bug
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| -- Executing [1001@sipusers:1] NoOp("SIP/intercom-test-00000137", "Intercom dialplan") in new stack | |
| -- Executing [1001@sipusers:2] Originate("SIP/intercom-test-00000137", "SIP/microsip-test,exten,intercom_context,s,1,40") in new stack | |
| Audio is at 10102 | |
| Video is at 192.168.0.1:11038 | |
| Adding codec alaw to SDP | |
| Adding video codec h263 to SDP | |
| Adding video codec h264 to SDP | |
| Adding video codec vp8 to SDP | |
| Adding non-codec 0x1 (telephone-event) to SDP | |
| Reliably Transmitting (no NAT) to 192.168.0.140:40082: | |
| INVITE sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp> | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: Asterisk PBX | |
| Date: Sun, 25 Feb 2018 00:45:10 GMT | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, -- Called microsip-test | |
| Retransmitting #1 (no NAT) to 192.168.0.140:40082: | |
| INVITE sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp> | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: Asterisk PBX | |
| Date: Sun, 25 Feb 2018 00:45:10 GMT | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESS | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 100 Trying | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| <-------------> | |
| --- (6 headers 0 lines) --- | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 180 Ringing | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| <-------------> | |
| --- (8 headers 0 lines) --- | |
| sip_route_dump: no route/path | |
| -- SIP/microsip-test-00000138 is ringing | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 180 Ringing | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| <-------------> | |
| --- (8 headers 0 lines) --- | |
| sip_route_dump: no route/path | |
| -- SIP/microsip-test-00000138 is ringing | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 200 Ok | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE | |
| Contact: <sip:microsip-test@192.168.0.140:40082;transport=udp>;+sip.instance="<urn:uuid:1b5e3430-85aa-4a15-8018-508b370672fe>" | |
| Content-Type: application/sdp | |
| Content-Length: 187 | |
| v=0 | |
| o=microsip-test 630 3022 IN IP4 192.168.0.140 | |
| s=Talk | |
| c=IN IP4 192.168.0.140 | |
| t=0 0 | |
| m=audio 7076 RTP/AVP 8 Ъ--- (12 headers 9 lines) --- | |
| Found RTP audio format 8 | |
| Found RTP audio format 101 | |
| Found audio description format telephone-event for ID 101 | |
| Capabilities: us - (alaw|h263|h264|vp8), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) | |
| Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) | |
| > 0x805d2d000 -- Strict RTP learning after remote address set to: 192.168.0.140:7076 | |
| Peer audio RTP is at port 192.168.0.140:7076 | |
| Peer doesn't provide video | |
| sip_route_dump: route/path hop: <sip:microsip-test@192.168.0.140:40082;transport=udp> | |
| set_destination: Parsing <sip:microsip-test@192.168.0.140:40082;transport=udp> for address/port to send to | |
| set_destination: set destination to 192.168.0.140:40082 | |
| Transmitting (no NAT) to 192.168.0.140:40082: | |
| ACK sip:microsip-test@192.168.0.140:40082;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK4c52a10c | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transpo | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| -- SIP/microsip-test-00000138 answered | |
| -- Executing [s@intercom_context:1] NoOp("SIP/microsip-test-00000138", "Human joins conference") in new stack | |
| -- Executing [s@intercom_context:2] ConfBridge("SIP/microsip-test-00000138", "111,intercom_bridge,human_user") in new stack | |
| -- Executing [1001@sipusers:3] ExecIf("SIP/intercom-test-00000137", "1?Answer:Hangup") in new stack | |
| Audio is at 11846 | |
| Adding codec alaw to SDP | |
| Adding non-codec 0x1 (telephone-event) to SDP | |
| <--- Reliably Transmitting (no NAT) to 192.168.0.3:5060 ---> | |
| SIP/2.0 200 OK | |
| Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bKPjc23d029fddcc4bc3b0281b7761c79384;received=192.168.0.3;rport=5060 | |
| From: <sip:intercom-test@192.168.0.1>;tag=fd8cb9723e2f4cbf878bfb602561fab9 | |
| To: <sip:1001@192.168.0.1>;tag=as2d42630b | |
| Call-ID: 9d8afd0a4e604014b670d01b9af696b6 | |
| CSeq: 13067 INVITE | |
| Server: Asterisk PBX | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE | |
| Supported: replaces, timer | |
| Session-Expires: 1800;refresher=uas | |
| Contact: <sip:1001@192.168.0.1:5060> | |
| Content-Type: application/sdp | |
| Require: timer | |
| Content-Length: 236 | |
| v=0 | |
| o=root 611372779 611372779 IN IP4 192.168.0.1 | |
| s=Asterisk PBX 13.18.3 | |
| c=IN IP4 192.168.0.1 | |
| t=0 0 | |
| m=audio 11846 RTP/AVP 8 101 | |
| a=rtpmap:8 PCMA/8000 | |
| a=rtpmap:101 telephone-event/8000 | |
| a=fmtp:101 0-16 | |
| a=maxptime:150 | |
| a=sendrecv | |
| <------------> | |
| -- <SIP/microsip-test-00000138> Playing 'confbridge-join.slin' (language 'ru') | |
| -- Channel CBAnn/111-00000049;2 joined 'softmix' base-bridge <d6c87bcb-f553-40f2-8d18-37b6b3099565> | |
| <--- SIP read from UDP:192.168.0.3:5060 ---> | |
| ACK sip:1001@192.168.0.1:5060 SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.3:5060;rport;branch=z9hG4bKPjc593bd4e60594cffa1de92e268d1fa07 | |
| Max-Forwards: 70 | |
| From: <sip:intercom-test@192.168.0.1>;tag=fd8cb9723e2f4cbf878bfb602561fab9 | |
| To: <sip:1001@192.168.0.1>;tag=as2d42630b | |
| Call-ID: 9d8afd0a4e604014b670d01b9af696b6 | |
| CSeq: 13067 ACK | |
| Content-Length: 0 | |
| <-------------> | |
| --- (8 headers 0 lines) --- | |
| > 0x805cd8000 -- Strict RTP switching to RTP target address 192.168.0.3:4120 as source | |
| -- Executing [1001@sipusers:4] Originate("SIP/intercom-test-00000137", "SIP/linphonec-test,exten,video_source_context,s,1,3") in new stack | |
| Audio is at 19526 | |
| Video is at 192.168.0.1:19982 | |
| Adding codec alaw to SDP | |
| Adding video codec h263 to SDP | |
| Adding video codec h264 to SDP | |
| Adding video codec vp8 to SDP | |
| Adding non-codec 0x1 (telephone-event) to SDP | |
| Reliably Transmitting (no NAT) to 192.168.0.117:5060: | |
| INVITE sip:linphonec-test@192.168.0.117;line=a7aa9575076c6e5 SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5770442f | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as38d9f894 | |
| To: <sip:linphonec-test@192.168.0.117;line=a7aa9575076c6e5> | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 18c9d6d73ca5493663418b343afdf437@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: Asterisk PBX | |
| Date: Sun, 25 Feb 2018 00:45:13 GMT | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE | |
| Supported: replaces, timer | |
| Content-Type: application/sdp | |
| Content-Length: 384 | |
| v=0 | |
| o=root 116230161 116230161 IN IP4 192.168.0.1 | |
| s=Asterisk PBX 13.18.3 | |
| c=IN IP4 192.168.0.1 | |
| b=CT:384 | |
| t=0 0 | |
| m=audio 19526 RTP/AVP 8 101 | |
| a=rtpmap:8 PCMA/8000 | |
| a=rtpmap:101 telephone-event/8000 | |
| a=fmtp:101 0-16 | |
| a=maxptime:150 | |
| a=sendrecv | |
| m=video 19982 RTP/AVP 34 99 100 | |
| a=rtpmap:34 H263/90000 | |
| a= -- Called linphonec-test | |
| <--- SIP read from UDP:192.168.0.117:5060 ---> | |
| SIP/2.0 100 Trying | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5770442f | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as38d9f894 | |
| To: <sip:linphonec-test@192.168.0.117;line=a7aa9575076c6e5> | |
| Call-ID: 18c9d6d73ca5493663418b343afdf437@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) | |
| Content-Length: 0 | |
| <-------------> | |
| --- (8 headers 0 lines) --- | |
| <--- SIP read from UDP:192.168.0.117:5060 ---> | |
| SIP/2.0 180 Ringing | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5770442f | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as38d9f894 | |
| To: <sip:linphonec-test@192.168.0.117;line=a7aa9575076c6e5>;tag=2144142251 | |
| Call-ID: 18c9d6d73ca5493663418b343afdf437@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| Contact: <sip:linphonec-test@192.168.0.117:5060> | |
| User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) | |
| Content-Length: 0 | |
| <-------------> | |
| --- (9 headers 0 lines) --- | |
| sip_route_dump: route/path hop: <sip:linphonec-test@192.168.0.117:5060> | |
| <--- SIP read from UDP:192.168.0.117:5060 ---> | |
| SIP/2.0 200 OK | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5770442f | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as38d9f894 | |
| To: <sip:linphonec-test@192.168.0.117;line=a7aa9575076c6e5>;tag=2144142251 | |
| Call-ID: 18c9d6d73ca5493663418b343afdf437@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| Contact: <sip:petruchito@192.168.0.117> | |
| Content-Type: application/sdp | |
| User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) | |
| Content-Length: 245 | |
| v=0 | |
| o=linphonec-test 3956 3349 IN IP4 192.168.0.117 | |
| s=Talk | |
| c=IN IP4 192.168.0.117 | |
| t=0 0 | |
| m=audio 7078 RTP/AVP 8 101 | |
| a=rtpmap:8 PCMA/8000 | |
| a=rtpmap:101 telephone-event/8000 | |
| a=fmtp:101 0-11 | |
| m=video 9078 RTP/AVP 100 | |
| a=rtpmap:100 VP8/90000 | |
| <-------------> | |
| --- (10 headers 11 lines) --- | |
| Found RTP audio format 8 | |
| Found RTP audio format 101 | |
| Found audio description format PCMA for ID 8 | |
| Found audio description format telephone-event for ID 101 | |
| Found RTP video format 100 | |
| Found video description format VP8 for ID 100 | |
| Capabilities: us - (alaw|h263|h264|vp8), peer - audio=(alaw)/video=(vp8)/text=(nothing), combined - (alaw|vp8) | |
| Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) | |
| > 0x805e72000 -- Strict RTP learning after remote address set to: 192.168.0.117:7078 | |
| Peer audio RTP is at port 192.168.0.117:7078 | |
| > 0x805e7d000 -- Strict RTP learning after remote address set to: 192.168.0.117:9078 | |
| Peer video RTP is at port 192.168.0.117:9078 | |
| sip_route_dump: route/path hop: <sip:petruchito@192.168.0.117> | |
| set_destination: Parsing <sip:petruchito@192.168.0.117> for address/port to send to | |
| set_destination: set destination to 192.168.0.117:5060 | |
| Transmitting (no NAT) to 192.168.0.117:5060: | |
| ACK sip:petruchito@192.168.0.117 SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK10b9897a | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as38d9f894 | |
| To: <sip:linphonec-test@192.168.0.117;line=a7aa9575076c6e5>;tag=2144142251 | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 18c9d6d73ca5493663418b343afdf437@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| > 0x805cd8000 -- Strict RTP learning complete - Locking on source address 192.168.0.3:4120 | |
| -- SIP/linphonec-test-00000139 is ringing | |
| -- SIP/linphonec-test-00000139 answered | |
| -- Executing [s@video_source_context:1] NoOp("SIP/linphonec-test-00000139", "Connecting video source") in new stack | |
| -- Executing [s@video_source_context:2] ConfBridge("SIP/linphonec-test-00000139", "111,intercom_bridge,video_source_user") in new stack | |
| -- Executing [1001@sipusers:5] ConfBridge("SIP/intercom-test-00000137", "111,intercom_bridge,intercom_user") in new stack | |
| -- <SIP/linphonec-test-00000139> Playing 'confbridge-join.slin' (language 'ru') | |
| -- <SIP/intercom-test-00000137> Playing 'confbridge-join.slin' (language 'ru') | |
| > 0x805e72000 -- Strict RTP switching to RTP target address 192.168.0.117:7078 as source | |
| -- <CBAnn/111-00000049;1> Playing 'confbridge-join.gsm' (language 'en') | |
| > 0x805d2d000 -- Strict RTP switching to RTP target address 192.168.0.140:7076 as source | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 200 Ok | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE | |
| Contact: <sip:microsip-test@192.168.0.140:40082;transport=udp>;+sip.instance="<urn:uuid:1b5e3430-85aa-4a15-8018-508b370672fe>" | |
| Content-Type: application/sdp | |
| Content-Length: 187 | |
| v=0 | |
| o=microsip-test 630 3022 IN IP4 192.168.0.140 | |
| s=Talk | |
| c=IN IP4 192.168.0.140 | |
| t=0 0 | |
| m=audio 7076 RTP/AVP 8 --- (12 headers 9 lines) --- | |
| set_destination: Parsing <sip:microsip-test@192.168.0.140:40082;transport=udp> for address/port to send to | |
| set_destination: set destination to 192.168.0.140:40082 | |
| Transmitting (no NAT) to 192.168.0.140:40082: | |
| ACK sip:microsip-test@192.168.0.140:40082;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1f079847 | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transpo | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| > 0x805e7d000 -- Strict RTP switching to RTP target address 192.168.0.117:9078 as source | |
| -- Channel SIP/microsip-test-00000138 joined 'softmix' base-bridge <d6c87bcb-f553-40f2-8d18-37b6b3099565> | |
| -- <CBAnn/111-00000049;1> Playing 'confbridge-join.gsm' (language 'en') | |
| > Video source in bridge '111' (d6c87bcb-f553-40f2-8d18-37b6b3099565) is now 'SIP/linphonec-test-00000139' (1519519513.866) | |
| -- Channel SIP/linphonec-test-00000139 joined 'softmix' base-bridge <d6c87bcb-f553-40f2-8d18-37b6b3099565> | |
| -- <CBAnn/111-00000049;1> Playing 'confbridge-join.gsm' (language 'en') | |
| <--- SIP read from UDP:192.168.0.117:5060 ---> | |
| INFO sip:anonymous@192.168.0.1:5060 SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.117:5060;rport;branch=z9hG4bK709035327 | |
| From: <sip:linphonec-test@192.168.0.117;line=a7aa9575076c6e5>;tag=2144142251 | |
| To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as38d9f894 | |
| Call-ID: 18c9d6d73ca5493663418b343afdf437@192.168.0.1:5060 | |
| CSeq: 2 INFO | |
| Contact: <sip:linphonec-test@192.168.0.117:5060> | |
| Content-Type: application/media_control+xml | |
| Max-Forwards: 70 | |
| User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) | |
| Content-Length: 185 | |
| <?xml version="1.0" encoding="utf-8" ?><media_control> <vc_primitive> <to_encoder> <picture_fast_update></picture_fast_update> </to_encoder> </vc_primitive></media_control> | |
| <-------------> | |
| --- (11 headers 1 lines) --- | |
| Receiving INFO! | |
| <--- Transmitting (no NAT) to 192.168.0.117:5060 ---> | |
| SIP/2.0 200 OK | |
| Via: SIP/2.0/UDP 192.168.0.117:5060;rport;branch=z9hG4bK709035327;received=192.168.0.117 | |
| From: <sip:linphonec-test@192.168.0.117;line=a7aa9575076c6e5>;tag=2144142251 | |
| To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as38d9f894 | |
| Call-ID: 18c9d6d73ca5493663418b343afdf437@192.168.0.1:5060 | |
| CSeq: 2 INFO | |
| Server: Asterisk PBX | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE | |
| Supported: replaces, timer | |
| Content-Length: 0 | |
| <------------> | |
| > 0x805d2d000 -- Strict RTP learning complete - Locking on source address 192.168.0.140:7076 | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 200 Ok | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE | |
| Contact: <sip:microsip-test@192.168.0.140:40082;transport=udp>;+sip.instance="<urn:uuid:1b5e3430-85aa-4a15-8018-508b370672fe>" | |
| Content-Type: application/sdp | |
| Content-Length: 187 | |
| v=0 | |
| o=microsip-test 630 3022 IN IP4 192.168.0.140 | |
| s=Talk | |
| c=IN IP4 192.168.0.140 | |
| t=0 0 | |
| m=audio 7076 RTP/AVP 8 --- (12 headers 9 lines) --- | |
| set_destination: Parsing <sip:microsip-test@192.168.0.140:40082;transport=udp> for address/port to send to | |
| set_destination: set destination to 192.168.0.140:40082 | |
| Transmitting (no NAT) to 192.168.0.140:40082: | |
| ACK sip:microsip-test@192.168.0.140:40082;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK742605e7 | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transpo | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| > 0x805e72000 -- Strict RTP learning complete - Locking on source address 192.168.0.117:7078 | |
| -- Channel SIP/intercom-test-00000137 joined 'softmix' base-bridge <d6c87bcb-f553-40f2-8d18-37b6b3099565> | |
| > 0x805e7d000 -- Strict RTP learning complete - Locking on source address 192.168.0.117:9078 | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| <-------------> | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 200 Ok | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE | |
| Contact: <sip:microsip-test@192.168.0.140:40082;transport=udp>;+sip.instance="<urn:uuid:1b5e3430-85aa-4a15-8018-508b370672fe>" | |
| Content-Type: application/sdp | |
| Content-Length: 187 | |
| v=0 | |
| o=microsip-test 630 3022 IN IP4 192.168.0.140 | |
| s=Talk | |
| c=IN IP4 192.168.0.140 | |
| t=0 0 | |
| m=audio 7076 RTP/AVP 8 Ъ--- (12 headers 9 lines) --- | |
| set_destination: Parsing <sip:microsip-test@192.168.0.140:40082;transport=udp> for address/port to send to | |
| set_destination: set destination to 192.168.0.140:40082 | |
| Transmitting (no NAT) to 192.168.0.140:40082: | |
| ACK sip:microsip-test@192.168.0.140:40082;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK15768d7f | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transpo | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| <--- SIP read from UDP:192.168.0.3:5060 ---> | |
| <-------------> | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 200 Ok | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE | |
| Contact: <sip:microsip-test@192.168.0.140:40082;transport=udp>;+sip.instance="<urn:uuid:1b5e3430-85aa-4a15-8018-508b370672fe>" | |
| Content-Type: application/sdp | |
| Content-Length: 187 | |
| v=0 | |
| o=microsip-test 630 3022 IN IP4 192.168.0.140 | |
| s=Talk | |
| c=IN IP4 192.168.0.140 | |
| t=0 0 | |
| m=audio 7076 RTP/AVP 8 Ъ--- (12 headers 9 lines) --- | |
| set_destination: Parsing <sip:microsip-test@192.168.0.140:40082;transport=udp> for address/port to send to | |
| set_destination: set destination to 192.168.0.140:40082 | |
| Transmitting (no NAT) to 192.168.0.140:40082: | |
| ACK sip:microsip-test@192.168.0.140:40082;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5558bba2 | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transpo | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 200 Ok | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE | |
| Contact: <sip:microsip-test@192.168.0.140:40082;transport=udp>;+sip.instance="<urn:uuid:1b5e3430-85aa-4a15-8018-508b370672fe>" | |
| Content-Type: application/sdp | |
| Content-Length: 187 | |
| v=0 | |
| o=microsip-test 630 3022 IN IP4 192.168.0.140 | |
| s=Talk | |
| c=IN IP4 192.168.0.140 | |
| t=0 0 | |
| m=audio 7076 RTP/AVP 8 Ъ--- (12 headers 9 lines) --- | |
| set_destination: Parsing <sip:microsip-test@192.168.0.140:40082;transport=udp> for address/port to send to | |
| set_destination: set destination to 192.168.0.140:40082 | |
| Transmitting (no NAT) to 192.168.0.140:40082: | |
| ACK sip:microsip-test@192.168.0.140:40082;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31859524 | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transpo | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 200 Ok | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE | |
| Contact: <sip:microsip-test@192.168.0.140:40082;transport=udp>;+sip.instance="<urn:uuid:1b5e3430-85aa-4a15-8018-508b370672fe>" | |
| Content-Type: application/sdp | |
| Content-Length: 187 | |
| v=0 | |
| o=microsip-test 630 3022 IN IP4 192.168.0.140 | |
| s=Talk | |
| c=IN IP4 192.168.0.140 | |
| t=0 0 | |
| m=audio 7076 RTP/AVP 8 Ъ--- (12 headers 9 lines) --- | |
| set_destination: Parsing <sip:microsip-test@192.168.0.140:40082;transport=udp> for address/port to send to | |
| set_destination: set destination to 192.168.0.140:40082 | |
| Transmitting (no NAT) to 192.168.0.140:40082: | |
| ACK sip:microsip-test@192.168.0.140:40082;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK59b3adb5 | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transpo | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| <--- SIP read from UDP:192.168.0.117:5060 ---> | |
| jaK | |
| <-------------> | |
| <--- SIP read from UDP:192.168.0.3:5060 ---> | |
| <-------------> | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 200 Ok | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE | |
| Contact: <sip:microsip-test@192.168.0.140:40082;transport=udp>;+sip.instance="<urn:uuid:1b5e3430-85aa-4a15-8018-508b370672fe>" | |
| Content-Type: application/sdp | |
| Content-Length: 187 | |
| v=0 | |
| o=microsip-test 630 3022 IN IP4 192.168.0.140 | |
| s=Talk | |
| c=IN IP4 192.168.0.140 | |
| t=0 0 | |
| m=audio 7076 RTP/AVP 8 Ъ--- (12 headers 9 lines) --- | |
| set_destination: Parsing <sip:microsip-test@192.168.0.140:40082;transport=udp> for address/port to send to | |
| set_destination: set destination to 192.168.0.140:40082 | |
| Transmitting (no NAT) to 192.168.0.140:40082: | |
| ACK sip:microsip-test@192.168.0.140:40082;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK037f713d | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transpo | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 200 Ok | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE | |
| Contact: <sip:microsip-test@192.168.0.140:40082;transport=udp>;+sip.instance="<urn:uuid:1b5e3430-85aa-4a15-8018-508b370672fe>" | |
| Content-Type: application/sdp | |
| Content-Length: 187 | |
| v=0 | |
| o=microsip-test 630 3022 IN IP4 192.168.0.140 | |
| s=Talk | |
| c=IN IP4 192.168.0.140 | |
| t=0 0 | |
| m=audio 7076 RTP/AVP 8 Ъ--- (12 headers 9 lines) --- | |
| set_destination: Parsing <sip:microsip-test@192.168.0.140:40082;transport=udp> for address/port to send to | |
| set_destination: set destination to 192.168.0.140:40082 | |
| Transmitting (no NAT) to 192.168.0.140:40082: | |
| ACK sip:microsip-test@192.168.0.140:40082;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0380a7a5 | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transpo | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| <--- SIP read from UDP:192.168.0.117:5060 ---> | |
| jaK | |
| <-------------> | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 200 Ok | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE | |
| Contact: <sip:microsip-test@192.168.0.140:40082;transport=udp>;+sip.instance="<urn:uuid:1b5e3430-85aa-4a15-8018-508b370672fe>" | |
| Content-Type: application/sdp | |
| Content-Length: 187 | |
| v=0 | |
| o=microsip-test 630 3022 IN IP4 192.168.0.140 | |
| s=Talk | |
| c=IN IP4 192.168.0.140 | |
| t=0 0 | |
| m=audio 7076 RTP/AVP 8 Ъ--- (12 headers 9 lines) --- | |
| set_destination: Parsing <sip:microsip-test@192.168.0.140:40082;transport=udp> for address/port to send to | |
| set_destination: set destination to 192.168.0.140:40082 | |
| Transmitting (no NAT) to 192.168.0.140:40082: | |
| ACK sip:microsip-test@192.168.0.140:40082;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK51d3bfa9 | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transpo | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| SIP/2.0 200 Ok | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32f53de8 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transport=udp>;tag=dKr~r~y | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 INVITE | |
| User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3) | |
| Supported: replaces, outbound | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE | |
| Contact: <sip:microsip-test@192.168.0.140:40082;transport=udp>;+sip.instance="<urn:uuid:1b5e3430-85aa-4a15-8018-508b370672fe>" | |
| Content-Type: application/sdp | |
| Content-Length: 187 | |
| v=0 | |
| o=microsip-test 630 3022 IN IP4 192.168.0.140 | |
| s=Talk | |
| c=IN IP4 192.168.0.140 | |
| t=0 0 | |
| m=audio 7076 RTP/AVP 8 Ъ--- (12 headers 9 lines) --- | |
| set_destination: Parsing <sip:microsip-test@192.168.0.140:40082;transport=udp> for address/port to send to | |
| set_destination: set destination to 192.168.0.140:40082 | |
| Transmitting (no NAT) to 192.168.0.140:40082: | |
| ACK sip:microsip-test@192.168.0.140:40082;transport=udp SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0259a03d | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| To: <sip:microsip-test@192.168.0.140:40082;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1;transpo | |
| Contact: <sip:anonymous@192.168.0.1:5060> | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 102 ACK | |
| User-Agent: Asterisk PBX | |
| Content-Length: 0 | |
| --- | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| <-------------> | |
| <--- SIP read from UDP:192.168.0.140:40082 ---> | |
| BYE sip:anonymous@192.168.0.1:5060 SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.140:40082;branch=z9hG4bK.r5UZH77Pz;rport | |
| From: <sip:microsip-test@192.168.0.140;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1>;tag=dKr~r~y | |
| To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| CSeq: 111 BYE | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| Max-Forwards: 70 | |
| <-------------> | |
| --- (7 headers 0 lines) --- | |
| Sending to 192.168.0.140:40082 (no NAT) | |
| Scheduling destruction of SIP dialog '22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060' in 32000 ms (Method: BYE) | |
| <--- Transmitting (no NAT) to 192.168.0.140:40082 ---> | |
| SIP/2.0 200 OK | |
| Via: SIP/2.0/UDP 192.168.0.140:40082;branch=z9hG4bK.r5UZH77Pz;received=192.168.0.140;rport=40082 | |
| From: <sip:microsip-test@192.168.0.140;app-id=929724111839;pn-type=firebase;pn-tok=cjmSEanoeVc:APA91bE3RpqAVHBcY8Fygdmnn6Xq9RqmJ1xXNK8RysMbrp4VbUE7odlVk6vkXEzEk3hdu3RFXlAsawkRsTflFRJsq46wE-wtOHvvKHTOf65g-ucpmchfkvKRBQUAN6acTZVr33dmokCU;pn-silent=1>;tag=dKr~r~y | |
| To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as41aeb74a | |
| Call-ID: 22da32872e53e50556bc6c4c4eb67cf8@192.168.0.1:5060 | |
| CSeq: 111 BYE | |
| Server: Asterisk PBX | |
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE | |
| Supported: replaces, timer | |
| Content-Length: 0 | |
| <------------> | |
| -- Channel SIP/microsip-test-00000138 left 'softmix' base-bridge <d6c87bcb-f553-40f2-8d18-37b6b3099565> | |
| -- Executing [h@intercom_context:1] NoOp("SIP/microsip-test-00000138", "Kick all from conference") in new stack | |
| -- Executing [h@intercom_context:2] System("SIP/microsip-test-00000138", "asterisk -rx "confbridge kick 111 all"") in new stack | |
| -- <CBAnn/111-00000049;1> Playing 'confbridge-leave.gsm' (language 'en') | |
| -- Remote UNIX connection | |
| -- Channel SIP/linphonec-test-00000139 left 'softmix' base-bridge <d6c87bcb-f553-40f2-8d18-37b6b3099565> | |
| -- Channel SIP/intercom-test-00000137 left 'softmix' base-bridge <d6c87bcb-f553-40f2-8d18-37b6b3099565> | |
| -- Remote UNIX connection disconnected | |
| -- <SIP/linphonec-test-00000139> Playing 'conf-kicked.slin' (language 'ru') | |
| -- <SIP/intercom-test-00000137> Playing 'conf-kicked.slin' (language 'ru') | |
| -- <CBAnn/111-00000049;1> Playing 'confbridge-leave.gsm' (language 'en') | |
| -- <CBAnn/111-00000049;1> Playing 'confbridge-leave.gsm' (language 'en') | |
| <--- SIP read from UDP:192.168.0.3:5060 ---> | |
| <-------------> | |
| -- Auto fallthrough, channel 'SIP/linphonec-test-00000139' status is 'UNKNOWN' | |
| Scheduling destruction of SIP dialog '18c9d6d73ca5493663418b343afdf437@192.168.0.1:5060' in 32000 ms (Method: INFO) | |
| set_destination: Parsing <sip:petruchito@192.168.0.117> for address/port to send to | |
| set_destination: set destination to 192.168.0.117:5060 | |
| Reliably Transmitting (no NAT) to 192.168.0.117:5060: | |
| BYE sip:petruchito@192.168.0.117 SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5072b175 | |
| Max-Forwards: 70 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as38d9f894 | |
| To: <sip:linphonec-test@192.168.0.117;line=a7aa9575076c6e5>;tag=2144142251 | |
| Call-ID: 18c9d6d73ca5493663418b343afdf437@192.168.0.1:5060 | |
| CSeq: 103 BYE | |
| User-Agent: Asterisk PBX | |
| X-Asterisk-HangupCause: Normal Clearing | |
| X-Asterisk-HangupCauseCode: 16 | |
| Content-Length: 0 | |
| --- | |
| <--- SIP read from UDP:192.168.0.117:5060 ---> | |
| SIP/2.0 200 OK | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5072b175 | |
| From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as38d9f894 | |
| To: <sip:linphonec-test@192.168.0.117;line=a7aa9575076c6e5>;tag=2144142251 | |
| Call-ID: 18c9d6d73ca5493663418b343afdf437@192.168.0.1:5060 | |
| CSeq: 103 BYE | |
| User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) | |
| Content-Length: 0 | |
| <-------------> | |
| --- (8 headers 0 lines) --- | |
| Really destroying SIP dialog '18c9d6d73ca5493663418b343afdf437@192.168.0.1:5060' Method: INFO | |
| -- <CBAnn/111-00000049;1> Playing 'conf-leaderhasleft.gsm' (language 'en') | |
| -- Channel CBAnn/111-00000049;2 left 'softmix' base-bridge <d6c87bcb-f553-40f2-8d18-37b6b3099565> | |
| -- Auto fallthrough, channel 'SIP/intercom-test-00000137' status is 'UNKNOWN' | |
| Scheduling destruction of SIP dialog '9d8afd0a4e604014b670d01b9af696b6' in 32000 ms (Method: ACK) | |
| set_destination: Parsing <sip:intercom-test@192.168.0.3:5060;ob> for address/port to send to | |
| set_destination: set destination to 192.168.0.3:5060 | |
| Reliably Transmitting (no NAT) to 192.168.0.3:5060: | |
| BYE sip:intercom-test@192.168.0.3:5060;ob SIP/2.0 | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK137ffa4d;rport | |
| Max-Forwards: 70 | |
| From: <sip:1001@192.168.0.1>;tag=as2d42630b | |
| To: <sip:intercom-test@192.168.0.1>;tag=fd8cb9723e2f4cbf878bfb602561fab9 | |
| Call-ID: 9d8afd0a4e604014b670d01b9af696b6 | |
| CSeq: 102 BYE | |
| User-Agent: Asterisk PBX | |
| Proxy-Authorization: Digest username="intercom-test", realm="asterisk", algorithm=MD5, uri="sip:192.168.0.1", nonce="4c9a7dac", response="3beb7f0d716f3dc0a8eeea26b72515f3" | |
| X-Asterisk-HangupCause: Normal Clearing | |
| X-Asterisk-HangupCauseCode: 16 | |
| Content-Length: 0 | |
| --- | |
| <--- SIP read from UDP:192.168.0.3:5060 ---> | |
| SIP/2.0 200 OK | |
| Via: SIP/2.0/UDP 192.168.0.1:5060;rport=5060;received=192.168.0.1;branch=z9hG4bK137ffa4d | |
| Call-ID: 9d8afd0a4e604014b670d01b9af696b6 | |
| From: <sip:1001@192.168.0.1>;tag=as2d42630b | |
| To: <sip:intercom-test@192.168.0.1>;tag=fd8cb9723e2f4cbf878bfb602561fab9 | |
| CSeq: 102 BYE | |
| Content-Length: 0 | |
| <-------------> | |
| --- (7 headers 0 lines) --- | |
| SIP Response message for INCOMING dialog BYE arrived | |
| Really destroying SIP dialog '9d8afd0a4e604014b670d01b9af696b6' Method: ACK | |
| router*CLI> sip set debug off |
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