[Oct 18 10:02:02] Asterisk 14.0.2 built by root @ debian on a i686 running Linux on 2016-10-13 14:49:59 UTC [Oct 18 10:02:02] DEBUG[23593] config.c: Parsing /home/user/projects/asterisk_test/configs/minimal/logger.conf [Oct 18 10:02:02] VERBOSE[23593] config.c: Parsing '/home/user/projects/asterisk_test/configs/minimal/logger.conf': Found [Oct 18 10:02:02] VERBOSE[23593] logger.c: Asterisk Queue Logger restarted [Oct 18 10:02:11] DEBUG[23625] chan_sip.c: = Looking for Call ID: ZTb0a8rVjorgQA-7U.zzLouKvspcQHJP (Checking From) --From tag pCtxg9Jm4.-5hcSBFAh7v1Fig9hbo7K9 --To-tag [Oct 18 10:02:11] DEBUG[23625] acl.c: For destination '192.168.210.110', our source address is '192.168.210.71'. [Oct 18 10:02:11] DEBUG[23625] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.210.71:5060 [Oct 18 10:02:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:02:11] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:02:11] DEBUG[23625] chan_sip.c: Allocating new SIP dialog for ZTb0a8rVjorgQA-7U.zzLouKvspcQHJP - SUBSCRIBE (No RTP) [Oct 18 10:02:11] DEBUG[23625] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 18 10:02:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:02:11] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:02:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:02:11] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:02:11] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.210.110:53027 [Oct 18 10:02:11] DEBUG[23625] chan_sip.c: = Looking for Call ID: ZTb0a8rVjorgQA-7U.zzLouKvspcQHJP (Checking From) --From tag pCtxg9Jm4.-5hcSBFAh7v1Fig9hbo7K9 --To-tag [Oct 18 10:02:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:02:11] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:02:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:02:11] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:02:11] DEBUG[23625] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 18 10:02:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:02:11] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:02:11] DEBUG[23625] chan_sip.c: build_route: Retaining previous route: [Oct 18 10:02:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:02:11] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:02:11] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 192.168.210.110:53027 [Oct 18 10:02:11] NOTICE[23625] chan_sip.c: Received SIP subscribe for peer without mailbox: 300 [Oct 18 10:02:11] DEBUG[23625] chan_sip.c: Destroying SIP dialog ZTb0a8rVjorgQA-7U.zzLouKvspcQHJP [Oct 18 10:02:22] DEBUG[23645] http.c: HTTP opening session. Top level [Oct 18 10:02:22] DEBUG[23645] http.c: HTTP Request URI is /ari/events?api_key=asterisk:asterisk&app=attendant&subscribeAll=true [Oct 18 10:02:22] DEBUG[23645] http.c: match request [ari/events] with handler [httpstatus] len 10 [Oct 18 10:02:22] DEBUG[23645] http.c: match request [ari/events] with handler [amanager] len 8 [Oct 18 10:02:22] DEBUG[23645] http.c: match request [ari/events] with handler [arawman] len 7 [Oct 18 10:02:22] DEBUG[23645] http.c: match request [ari/events] with handler [manager] len 7 [Oct 18 10:02:22] DEBUG[23645] http.c: match request [ari/events] with handler [rawman] len 6 [Oct 18 10:02:22] DEBUG[23645] http.c: match request [ari/events] with handler [static] len 6 [Oct 18 10:02:22] DEBUG[23645] http.c: match request [ari/events] with handler [amxml] len 5 [Oct 18 10:02:22] DEBUG[23645] http.c: match request [ari/events] with handler [mxml] len 4 [Oct 18 10:02:22] DEBUG[23645] http.c: match request [ari/events] with handler [ari] len 3 [Oct 18 10:02:22] DEBUG[23645] http.c: Match made with [ari] [Oct 18 10:02:22] DEBUG[23645] res_ari.c: Finding handler for events [Oct 18 10:02:22] DEBUG[23645] res_ari.c: Checking events [Oct 18 10:02:22] DEBUG[23645] res_ari.c: Got it! [Oct 18 10:02:22] DEBUG[23645] ari/resource_events.c: /events WebSocket attempted [Oct 18 10:02:22] VERBOSE[23645] stasis/app.c: Creating Stasis app 'attendant' [Oct 18 10:02:22] DEBUG[23645] stasis/app.c: Channel 'ALL' is 1 interested in attendant [Oct 18 10:02:22] DEBUG[23645] stasis/app.c: Bridge 'ALL' is 1 interested in attendant [Oct 18 10:02:22] DEBUG[23645] stasis/messaging.c: App 'attendant' subscribed to messages from endpoint '-- ALL --' [Oct 18 10:02:22] DEBUG[23645] stasis/app.c: Endpoint 'ALL' is 1 interested in attendant [Oct 18 10:02:22] DEBUG[23645] res_stasis_device_state.c: Subscribing to device __AST_DEVICE_STATE_ALL_TOPIC [Oct 18 10:02:22] DEBUG[23596] threadpool.c: Increasing threadpool stasis-core's size by 1 [Oct 18 10:02:22] VERBOSE[23645] res_http_websocket.c: WebSocket connection from '127.0.0.1:33266' for protocol '' accepted using version '13' [Oct 18 10:02:22] DEBUG[23645] ari/resource_events.c: /events WebSocket established [Oct 18 10:02:29] DEBUG[23652] http.c: HTTP opening session. Top level [Oct 18 10:02:29] DEBUG[23652] http.c: HTTP Request URI is /ari/applications/attendant/subscription?eventSource=channel:,endpoint:,bridge:,deviceState: [Oct 18 10:02:29] DEBUG[23652] http.c: match request [ari/applications/attendant/subscription] with handler [httpstatus] len 10 [Oct 18 10:02:29] DEBUG[23652] http.c: match request [ari/applications/attendant/subscription] with handler [amanager] len 8 [Oct 18 10:02:29] DEBUG[23652] http.c: match request [ari/applications/attendant/subscription] with handler [arawman] len 7 [Oct 18 10:02:29] DEBUG[23652] http.c: match request [ari/applications/attendant/subscription] with handler [manager] len 7 [Oct 18 10:02:29] DEBUG[23652] http.c: match request [ari/applications/attendant/subscription] with handler [rawman] len 6 [Oct 18 10:02:29] DEBUG[23652] http.c: match request [ari/applications/attendant/subscription] with handler [static] len 6 [Oct 18 10:02:29] DEBUG[23652] http.c: match request [ari/applications/attendant/subscription] with handler [amxml] len 5 [Oct 18 10:02:29] DEBUG[23652] http.c: match request [ari/applications/attendant/subscription] with handler [mxml] len 4 [Oct 18 10:02:29] DEBUG[23652] http.c: match request [ari/applications/attendant/subscription] with handler [ari] len 3 [Oct 18 10:02:29] DEBUG[23652] http.c: Match made with [ari] [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Finding handler for applications [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Checking events [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Checking bridges [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Checking channels [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Checking asterisk [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Checking applications [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Got it! [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Finding handler for attendant [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Checking applicationName [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Got it! [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Finding handler for subscription [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Checking subscription [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Got it! [Oct 18 10:02:29] DEBUG[23652] res_stasis.c: attendant: Checking channel: [Oct 18 10:02:29] DEBUG[23652] res_stasis.c: attendant: Subscribing to channel: [Oct 18 10:02:29] DEBUG[23652] res_stasis.c: attendant: Checking endpoint: [Oct 18 10:02:29] DEBUG[23652] res_stasis.c: attendant: Subscribing to endpoint: [Oct 18 10:02:29] DEBUG[23652] res_stasis.c: attendant: Checking bridge: [Oct 18 10:02:29] DEBUG[23652] res_stasis.c: attendant: Subscribing to bridge: [Oct 18 10:02:29] DEBUG[23652] res_stasis.c: attendant: Checking deviceState: [Oct 18 10:02:29] DEBUG[23652] res_stasis.c: attendant: Subscribing to deviceState: [Oct 18 10:02:29] DEBUG[23652] res_stasis_device_state.c: App attendant is already subscribed to __AST_DEVICE_STATE_ALL_TOPIC [Oct 18 10:02:29] DEBUG[23652] res_stasis.c: attendant: Successful; setting results [Oct 18 10:02:29] DEBUG[23652] res_ari.c: Examining ARI response: 200 OK Content-type: application/json { "name": "attendant", "channel_ids": [ "__AST_CHANNEL_ALL_TOPIC" ], "bridge_ids": [ "__AST_BRIDGE_ALL_TOPIC" ], "endpoint_ids": [ "__AST_ENDPOINT_ALL_TOPIC" ], "device_names": [ "__AST_DEVICE_STATE_ALL_TOPIC" ] } [Oct 18 10:02:29] DEBUG[23652] http.c: HTTP keeping session open. status_code:200 [Oct 18 10:02:29] DEBUG[23652] http.c: HTTP idle timeout or peer closed connection. [Oct 18 10:02:29] DEBUG[23652] http.c: HTTP closing session. Top level [Oct 18 10:02:35] DEBUG[23656] http.c: HTTP opening session. Top level [Oct 18 10:02:35] DEBUG[23656] http.c: HTTP Request URI is /ari/bridges [Oct 18 10:02:35] DEBUG[23656] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 18 10:02:35] DEBUG[23656] http.c: match request [ari/bridges] with handler [amanager] len 8 [Oct 18 10:02:35] DEBUG[23656] http.c: match request [ari/bridges] with handler [arawman] len 7 [Oct 18 10:02:35] DEBUG[23656] http.c: match request [ari/bridges] with handler [manager] len 7 [Oct 18 10:02:35] DEBUG[23656] http.c: match request [ari/bridges] with handler [rawman] len 6 [Oct 18 10:02:35] DEBUG[23656] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 18 10:02:35] DEBUG[23656] http.c: match request [ari/bridges] with handler [amxml] len 5 [Oct 18 10:02:35] DEBUG[23656] http.c: match request [ari/bridges] with handler [mxml] len 4 [Oct 18 10:02:35] DEBUG[23656] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 18 10:02:35] DEBUG[23656] http.c: Match made with [ari] [Oct 18 10:02:35] DEBUG[23656] res_ari.c: Finding handler for bridges [Oct 18 10:02:35] DEBUG[23656] res_ari.c: Checking events [Oct 18 10:02:35] DEBUG[23656] res_ari.c: Checking bridges [Oct 18 10:02:35] DEBUG[23656] res_ari.c: Got it! [Oct 18 10:02:35] DEBUG[23656] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 18 10:02:35] DEBUG[23656] bridge_native_rtp.c: Bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5' can not use native RTP bridge as two channels are required [Oct 18 10:02:35] DEBUG[23656] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 18 10:02:35] DEBUG[23656] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Oct 18 10:02:35] DEBUG[23656] bridge.c: Chose bridge technology simple_bridge [Oct 18 10:02:35] DEBUG[23656] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling simple_bridge technology constructor [Oct 18 10:02:35] DEBUG[23656] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling simple_bridge technology start [Oct 18 10:02:35] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 329): { "type": "BridgeCreated", "timestamp": "2016-10-18T10:02:35.556+0200", "bridge": { "id": "b1ecceb5-0432-40e6-b8c4-04dda2246cd5", "technology": "simple_bridge", "bridge_type": "mixing", "bridge_class": "stasis", "creator": "Stasis", "name": "", "channels": [] }, "application": "attendant" } [Oct 18 10:02:35] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 329 [Oct 18 10:02:35] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 329 [Oct 18 10:02:35] DEBUG[23656] res_ari.c: Examining ARI response: 200 OK Content-type: application/json { "id": "b1ecceb5-0432-40e6-b8c4-04dda2246cd5", "technology": "simple_bridge", "bridge_type": "mixing", "bridge_class": "stasis", "creator": "Stasis", "name": "", "channels": [] } [Oct 18 10:02:35] DEBUG[23656] http.c: HTTP keeping session open. status_code:200 [Oct 18 10:02:35] DEBUG[23656] http.c: HTTP idle timeout or peer closed connection. [Oct 18 10:02:35] DEBUG[23656] http.c: HTTP closing session. Top level [Oct 18 10:02:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:02:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:02:42] DEBUG[23649] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 18 10:02:42] DEBUG[23596] threadpool.c: Destroying worker thread 5 [Oct 18 10:02:54] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> <-------------> [Oct 18 10:02:58] DEBUG[23659] http.c: HTTP opening session. Top level [Oct 18 10:02:58] DEBUG[23659] http.c: HTTP Request URI is /ari/channels?endpoint=SIP/101&app=attendant [Oct 18 10:02:58] DEBUG[23659] http.c: match request [ari/channels] with handler [httpstatus] len 10 [Oct 18 10:02:58] DEBUG[23659] http.c: match request [ari/channels] with handler [amanager] len 8 [Oct 18 10:02:58] DEBUG[23659] http.c: match request [ari/channels] with handler [arawman] len 7 [Oct 18 10:02:58] DEBUG[23659] http.c: match request [ari/channels] with handler [manager] len 7 [Oct 18 10:02:58] DEBUG[23659] http.c: match request [ari/channels] with handler [rawman] len 6 [Oct 18 10:02:58] DEBUG[23659] http.c: match request [ari/channels] with handler [static] len 6 [Oct 18 10:02:58] DEBUG[23659] http.c: match request [ari/channels] with handler [amxml] len 5 [Oct 18 10:02:58] DEBUG[23659] http.c: match request [ari/channels] with handler [mxml] len 4 [Oct 18 10:02:58] DEBUG[23659] http.c: match request [ari/channels] with handler [ari] len 3 [Oct 18 10:02:58] DEBUG[23659] http.c: Match made with [ari] [Oct 18 10:02:58] DEBUG[23659] res_ari.c: Finding handler for channels [Oct 18 10:02:58] DEBUG[23659] res_ari.c: Checking events [Oct 18 10:02:58] DEBUG[23659] res_ari.c: Checking bridges [Oct 18 10:02:58] DEBUG[23659] res_ari.c: Checking channels [Oct 18 10:02:58] DEBUG[23659] res_ari.c: Got it! [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: Asked to create a SIP channel with formats: (slin) [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: Allocating new SIP dialog for 36811a314365f2f571269fef2d17a441@127.0.1.1:5060 - INVITE (No RTP) [Oct 18 10:02:58] DEBUG[23659] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb663a450' [Oct 18 10:02:58] DEBUG[23659] res_rtp_asterisk.c: Allocated port 8570 for RTP instance '0xb663a450' [Oct 18 10:02:58] DEBUG[23659] res_rtp_asterisk.c: Creating ICE session 0.0.0.0:8570 (8570) for RTP instance '0xb663a450' [Oct 18 10:02:58] DEBUG[23659] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:02:58] DEBUG[23659] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:02:58] DEBUG[23659] rtp_engine.c: RTP instance '0xb663a450' is setup and ready to go [Oct 18 10:02:58] DEBUG[23659] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb666b8d0' [Oct 18 10:02:58] DEBUG[23659] res_rtp_asterisk.c: Allocated port 27576 for RTP instance '0xb666b8d0' [Oct 18 10:02:58] DEBUG[23659] res_rtp_asterisk.c: Creating ICE session 0.0.0.0:27576 (27576) for RTP instance '0xb666b8d0' [Oct 18 10:02:58] DEBUG[23659] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:02:58] DEBUG[23659] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:02:58] DEBUG[23659] rtp_engine.c: RTP instance '0xb666b8d0' is setup and ready to go [Oct 18 10:02:58] DEBUG[23659] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb666b8d0' [Oct 18 10:02:58] DEBUG[23659] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb663a450' [Oct 18 10:02:58] VERBOSE[23659] netsock2.c: Using SIP RTP CoS mark 5 [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: Setting NAT on RTP to On [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: Setting NAT on TRTP to On [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Oct 18 10:02:58] DEBUG[23659] acl.c: For destination '192.168.210.40', our source address is '192.168.210.71'. [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.210.71:5060 [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: Setting NAT on RTP to On [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: Setting NAT on TRTP to On [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: SIP call-id changed from '36811a314365f2f571269fef2d17a441@127.0.1.1:5060' to '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' [Oct 18 10:02:58] DEBUG[23596] threadpool.c: Increasing threadpool stasis-core's size by 1 [Oct 18 10:02:58] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 534): { "type": "ChannelCreated", "timestamp": "2016-10-18T10:02:58.554+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Down", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "s", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 534 [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 534 [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: *** Our native formats are (alaw) [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: *** Joint capabilities are (nothing) [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: *** Our capabilities are (alaw) [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: *** Our preferred formats from the incoming channel are (slin) [Oct 18 10:02:58] DEBUG[23659] chan_sip.c: This channel will not be able to handle video. [Oct 18 10:02:58] DEBUG[23659] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 18 10:02:58] DEBUG[23659] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 18 10:02:58] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 627): { "value": "1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060", "variable": "SIPCALLID", "type": "ChannelVarset", "timestamp": "2016-10-18T10:02:58.556+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Down", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "s", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 627 [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 627 [Oct 18 10:02:58] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 606): { "type": "ChannelDialplan", "timestamp": "2016-10-18T10:02:58.557+0200", "dialplan_app": "AppDial2", "dialplan_app_data": "(Outgoing Line)", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Down", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 606 [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 606 [Oct 18 10:02:58] DEBUG[23659] res_stasis.c: attendant: Subscribing to atom_asterisk-1476777778.0 [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Outgoing Call for 101 [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Updating call counter for outgoing call [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Call to peer '101' is 1 out of 2147483647 [Oct 18 10:02:58] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 101 [Oct 18 10:02:58] DEBUG[23607] chan_sip.c: Checking device state for peer 101 [Oct 18 10:02:58] DEBUG[23607] devicestate.c: Changing state for SIP/101 - state 6 (Ringing) [Oct 18 10:02:58] DEBUG[23612] app_queue.c: Extension '101@LocalSets' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Oct 18 10:02:58] DEBUG[23635] app_queue.c: Device 'SIP/101' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Oct 18 10:02:58] DEBUG[23659] res_ari.c: Examining ARI response: 200 OK Content-type: application/json { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Down", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" } [Oct 18 10:02:58] DEBUG[23660] ari/ari_websockets.c: Examining ARI event (length 182): { "type": "DeviceStateChanged", "application": "attendant", "timestamp": "2016-10-18T10:02:58.560+0200", "device_state": { "name": "SIP/101", "state": "RINGING" } } [Oct 18 10:02:58] DEBUG[23659] http.c: HTTP keeping session open. status_code:200 [Oct 18 10:02:58] DEBUG[23660] res_http_websocket.c: Writing websocket string of length 182 [Oct 18 10:02:58] DEBUG[23660] res_http_websocket.c: Writing websocket text frame, length 182 [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: ** Our capability: (alaw) Video flag: False Text flag: False [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: ** Our prefcodec: (slin) [Oct 18 10:02:58] VERBOSE[23661] chan_sip.c: Audio is at 8570 [Oct 18 10:02:58] VERBOSE[23661] chan_sip.c: Adding codec alaw to SDP [Oct 18 10:02:58] VERBOSE[23661] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: -- Done with adding codecs to SDP [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Initializing initreq for method INVITE - callid 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 0 [ 42]: INVITE sip:101@192.168.210.40:5062 SIP/2.0 [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK0f3bda97;rport [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 3 [ 66]: From: "Anonymous" ;tag=as3af1af94 [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 4 [ 33]: To: [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 5 [ 44]: Contact: [Oct 18 10:02:58] DEBUG[23659] http.c: HTTP idle timeout or peer closed connection. [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 6 [ 61]: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:02:58] DEBUG[23659] http.c: HTTP closing session. Top level [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 14.0.2 [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 9 [ 35]: Date: Tue, 18 Oct 2016 08:02:58 GMT [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Oct 18 10:02:58] VERBOSE[23661] chan_sip.c: Reliably Transmitting (NAT) to 192.168.210.40:5062: INVITE sip:101@192.168.210.40:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK0f3bda97;rport Max-Forwards: 70 From: "Anonymous" ;tag=as3af1af94 To: Contact: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.0.2 Date: Tue, 18 Oct 2016 08:02:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 242 v=0 o=root 1361751049 1361751049 IN IP4 192.168.210.71 s=Asterisk PBX 14.0.2 c=IN IP4 192.168.210.71 t=0 0 m=audio 8570 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 --- [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #32 [Oct 18 10:02:58] DEBUG[23661] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.210.40:5062 [Oct 18 10:02:58] VERBOSE[23661] dial.c: Called 101 [Oct 18 10:02:58] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 580): { "type": "Dial", "timestamp": "2016-10-18T10:02:58.564+0200", "dialstatus": "", "forward": "", "dialstring": "101", "peer": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Down", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 580 [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 580 [Oct 18 10:02:58] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK0f3bda97;rport=5060 From: "Anonymous" ;tag=as3af1af94 To: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 102 INVITE User-Agent: Yealink SIP-T21P_E2 52.80.0.95 Content-Length: 0 <-------------> [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK0f3bda97;rport=5060 [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as3af1af94 [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 3 [ 33]: To: [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 6 [ 42]: User-Agent: Yealink SIP-T21P_E2 52.80.0.95 [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Oct 18 10:02:58] VERBOSE[23625] chan_sip.c: --- (8 headers 0 lines) --- [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: = Looking for Call ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 (Checking To) --From tag as3af1af94 --To-tag [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: *** SIP TIMER: Cancelling retransmission #32 - INVITE (got response) [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' Request 102: Found [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: SIP response 100 to standard invite [Oct 18 10:02:58] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK0f3bda97;rport=5060 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 102 INVITE Contact: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE User-Agent: Yealink SIP-T21P_E2 52.80.0.95 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 0 <-------------> [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK0f3bda97;rport=5060 [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as3af1af94 [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 3 [ 47]: To: ;tag=673443222 [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 6 [ 38]: Contact: [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 7 [115]: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 8 [ 42]: User-Agent: Yealink SIP-T21P_E2 52.80.0.95 [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 9 [ 51]: Allow-Events: talk,hold,conference,refer,check-sync [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Oct 18 10:02:58] VERBOSE[23625] chan_sip.c: --- (11 headers 0 lines) --- [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: = Looking for Call ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 (Checking To) --From tag as3af1af94 --To-tag 673443222 [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' Request 102: Found [Oct 18 10:02:58] DEBUG[23625] chan_sip.c: SIP response 180 to standard invite [Oct 18 10:02:58] VERBOSE[23625] sip/route.c: sip_route_dump: route/path hop: [Oct 18 10:02:58] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 101 [Oct 18 10:02:58] DEBUG[23607] chan_sip.c: Checking device state for peer 101 [Oct 18 10:02:58] DEBUG[23607] devicestate.c: Changing state for SIP/101 - state 6 (Ringing) [Oct 18 10:02:58] VERBOSE[23661] dial.c: SIP/101-00000000 is ringing [Oct 18 10:02:58] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 540): { "type": "ChannelStateChange", "timestamp": "2016-10-18T10:02:58.612+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Ringing", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 540 [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 540 [Oct 18 10:02:58] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 590): { "type": "Dial", "timestamp": "2016-10-18T10:02:58.613+0200", "dialstatus": "RINGING", "forward": "", "dialstring": "101", "peer": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Ringing", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 590 [Oct 18 10:02:58] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 590 [Oct 18 10:02:59] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK0f3bda97;rport=5060 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE User-Agent: Yealink SIP-T21P_E2 52.80.0.95 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 215 v=0 o=- 20002 20002 IN IP4 192.168.210.40 s=SDP data c=IN IP4 192.168.210.40 t=0 0 m=audio 11788 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK0f3bda97;rport=5060 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as3af1af94 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 3 [ 47]: To: ;tag=673443222 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 6 [ 38]: Contact: [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 8 [115]: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 9 [ 42]: User-Agent: Yealink SIP-T21P_E2 52.80.0.95 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 10 [ 51]: Allow-Events: talk,hold,conference,refer,check-sync [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 11 [ 19]: Content-Length: 215 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Header 12 [ 0]: [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Body 0 [ 3]: v=0 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Body 1 [ 37]: o=- 20002 20002 IN IP4 192.168.210.40 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Body 2 [ 10]: s=SDP data [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.210.40 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Body 5 [ 27]: m=audio 11788 RTP/AVP 8 101 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Body 7 [ 10]: a=ptime:20 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Body 10 [ 10]: a=sendrecv [Oct 18 10:02:59] VERBOSE[23625] chan_sip.c: --- (12 headers 11 lines) --- [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: = Looking for Call ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 (Checking To) --From tag as3af1af94 --To-tag 673443222 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Acked pending invite 102 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Stopping retransmission on '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' of Request 102: Match Found [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: SIP response 200 to standard invite [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Processing session-level SDP o=- 20002 20002 IN IP4 192.168.210.40... OK. [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Processing session-level SDP s=SDP data... UNSUPPORTED OR FAILED. [Oct 18 10:02:59] DEBUG[23625] netsock2.c: Splitting '192.168.210.40' into... [Oct 18 10:02:59] DEBUG[23625] netsock2.c: ...host '192.168.210.40' and port ''. [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.210.40... OK. [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 18 10:02:59] VERBOSE[23625] chan_sip.c: Found RTP audio format 8 [Oct 18 10:02:59] DEBUG[23625] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb385047c [Oct 18 10:02:59] VERBOSE[23625] chan_sip.c: Found RTP audio format 101 [Oct 18 10:02:59] DEBUG[23625] rtp_engine.c: Setting tx payload type 101 based on m type on 0xb385047c [Oct 18 10:02:59] VERBOSE[23625] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 18 10:02:59] VERBOSE[23625] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 18 10:02:59] VERBOSE[23625] chan_sip.c: Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 18 10:02:59] VERBOSE[23625] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 18 10:02:59] DEBUG[23625] res_rtp_asterisk.c: Set role to CONTROLLING (0xb663a450) [Oct 18 10:02:59] DEBUG[23625] res_rtp_asterisk.c: Set role failed; no ice instance (0xb663a450) [Oct 18 10:02:59] DEBUG[23625] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb663a450' [Oct 18 10:02:59] VERBOSE[23625] chan_sip.c: Peer audio RTP is at port 192.168.210.40:11788 [Oct 18 10:02:59] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 8 (0xb6401f80) from 0xb385047c to 0xb663a5fc [Oct 18 10:02:59] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 101 (0x8bedf80) from 0xb385047c to 0xb663a5fc [Oct 18 10:02:59] DEBUG[23625] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb663a450' [Oct 18 10:02:59] DEBUG[23625] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb666b8d0' [Oct 18 10:02:59] VERBOSE[23625] chan_sip.c: Peer doesn't provide T.140 [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: We're settling with these formats: (alaw) [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: We have an owner, now see if we need to change this call [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Updating call counter for outgoing call [Oct 18 10:02:59] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 101 [Oct 18 10:02:59] DEBUG[23607] chan_sip.c: Checking device state for peer 101 [Oct 18 10:02:59] DEBUG[23607] devicestate.c: Changing state for SIP/101 - state 2 (In use) [Oct 18 10:02:59] DEBUG[23635] app_queue.c: Device 'SIP/101' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 18 10:02:59] DEBUG[23660] ari/ari_websockets.c: Examining ARI event (length 180): { "type": "DeviceStateChanged", "application": "attendant", "timestamp": "2016-10-18T10:02:59.480+0200", "device_state": { "name": "SIP/101", "state": "INUSE" } } [Oct 18 10:02:59] DEBUG[23660] res_http_websocket.c: Writing websocket string of length 180 [Oct 18 10:02:59] DEBUG[23660] res_http_websocket.c: Writing websocket text frame, length 180 [Oct 18 10:02:59] DEBUG[23612] app_queue.c: Extension '101@LocalSets' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 18 10:02:59] VERBOSE[23625] sip/route.c: sip_route_dump: route/path hop: [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Strict routing enforced for session 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:02:59] VERBOSE[23625] chan_sip.c: Transmitting (NAT) to 192.168.210.40:5062: ACK sip:101@192.168.210.40:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK1efcf641;rport Max-Forwards: 70 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Contact: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 14.0.2 Content-Length: 0 --- [Oct 18 10:02:59] DEBUG[23625] chan_sip.c: Trying to put 'ACK sip:101' onto UDP socket destined for 192.168.210.40:5062 [Oct 18 10:02:59] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 535): { "type": "ChannelStateChange", "timestamp": "2016-10-18T10:02:59.481+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:02:59] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 535 [Oct 18 10:02:59] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 535 [Oct 18 10:02:59] VERBOSE[23661] dial.c: SIP/101-00000000 answered [Oct 18 10:02:59] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 101 [Oct 18 10:02:59] DEBUG[23607] chan_sip.c: Checking device state for peer 101 [Oct 18 10:02:59] DEBUG[23607] devicestate.c: Changing state for SIP/101 - state 2 (In use) [Oct 18 10:02:59] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 584): { "type": "Dial", "timestamp": "2016-10-18T10:02:59.482+0200", "dialstatus": "ANSWER", "forward": "", "dialstring": "101", "peer": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:02:59] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 584 [Oct 18 10:02:59] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 584 [Oct 18 10:02:59] VERBOSE[23661] ari/resource_channels.c: Launching Stasis(attendant) on SIP/101-00000000 [Oct 18 10:02:59] DEBUG[23596] threadpool.c: Increasing threadpool stasis-core's size by 1 [Oct 18 10:02:59] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 575): { "value": "", "variable": "STASISSTATUS", "type": "ChannelVarset", "timestamp": "2016-10-18T10:02:59.484+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:02:59] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 575 [Oct 18 10:02:59] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 575 [Oct 18 10:02:59] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 542): { "type": "StasisStart", "timestamp": "2016-10-18T10:02:59.485+0200", "args": [], "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:02:59] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 542 [Oct 18 10:02:59] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 542 [Oct 18 10:02:59] DEBUG[23661] res_rtp_asterisk.c: 0xb665a2c0 -- Probation learning mode pass with source address 192.168.210.40:11788 [Oct 18 10:02:59] VERBOSE[23661] res_rtp_asterisk.c: 0xb665a2c0 -- Probation passed - setting RTP source address to 192.168.210.40:11788 [Oct 18 10:03:04] DEBUG[23661] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 18 10:03:04] DEBUG[23661] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:04] DEBUG[23661] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:04] DEBUG[23661] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:04] DEBUG[23661] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:04] DEBUG[23661] acl.c: Attached to given IP address [Oct 18 10:03:04] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:04] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:04] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:04] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:04] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:09] DEBUG[23661] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 18 10:03:09] DEBUG[23661] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:09] DEBUG[23661] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:09] DEBUG[23661] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:09] DEBUG[23661] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:09] DEBUG[23661] acl.c: Attached to given IP address [Oct 18 10:03:09] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:09] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:09] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:09] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:09] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:03:10] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:03:14] DEBUG[23661] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 18 10:03:14] DEBUG[23661] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:14] DEBUG[23661] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:14] DEBUG[23661] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:14] DEBUG[23661] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:14] DEBUG[23661] acl.c: Attached to given IP address [Oct 18 10:03:14] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:14] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:14] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:14] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:14] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:19] DEBUG[23662] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 18 10:03:19] DEBUG[23596] threadpool.c: Destroying worker thread 7 [Oct 18 10:03:19] DEBUG[23660] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 18 10:03:19] DEBUG[23596] threadpool.c: Destroying worker thread 6 [Oct 18 10:03:19] DEBUG[23661] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 18 10:03:19] DEBUG[23661] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:19] DEBUG[23661] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:19] DEBUG[23661] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:19] DEBUG[23661] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:19] DEBUG[23661] acl.c: Attached to given IP address [Oct 18 10:03:19] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:19] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:19] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:19] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:19] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:21] DEBUG[23665] http.c: HTTP opening session. Top level [Oct 18 10:03:21] DEBUG[23665] http.c: HTTP Request URI is /ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel?channel=atom_asterisk-1476777778.0 [Oct 18 10:03:21] DEBUG[23665] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [httpstatus] len 10 [Oct 18 10:03:21] DEBUG[23665] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [amanager] len 8 [Oct 18 10:03:21] DEBUG[23665] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [arawman] len 7 [Oct 18 10:03:21] DEBUG[23665] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [manager] len 7 [Oct 18 10:03:21] DEBUG[23665] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [rawman] len 6 [Oct 18 10:03:21] DEBUG[23665] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [static] len 6 [Oct 18 10:03:21] DEBUG[23665] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [amxml] len 5 [Oct 18 10:03:21] DEBUG[23665] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [mxml] len 4 [Oct 18 10:03:21] DEBUG[23665] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [ari] len 3 [Oct 18 10:03:21] DEBUG[23665] http.c: Match made with [ari] [Oct 18 10:03:21] DEBUG[23665] res_ari.c: Finding handler for bridges [Oct 18 10:03:21] DEBUG[23665] res_ari.c: Checking events [Oct 18 10:03:21] DEBUG[23665] res_ari.c: Checking bridges [Oct 18 10:03:21] DEBUG[23665] res_ari.c: Got it! [Oct 18 10:03:21] DEBUG[23665] res_ari.c: Finding handler for b1ecceb5-0432-40e6-b8c4-04dda2246cd5 [Oct 18 10:03:21] DEBUG[23665] res_ari.c: Checking bridgeId [Oct 18 10:03:21] DEBUG[23665] res_ari.c: Got it! [Oct 18 10:03:21] DEBUG[23665] res_ari.c: Finding handler for addChannel [Oct 18 10:03:21] DEBUG[23665] res_ari.c: Checking addChannel [Oct 18 10:03:21] DEBUG[23665] res_ari.c: Got it! [Oct 18 10:03:21] DEBUG[23665] stasis/control.c: atom_asterisk-1476777778.0: Sending channel add_to_bridge command [Oct 18 10:03:21] DEBUG[23661] bridge_roles.c: Roles did not exist on channel SIP/101-00000000 [Oct 18 10:03:21] DEBUG[23661] stasis/control.c: atom_asterisk-1476777778.0: Adding to bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5 [Oct 18 10:03:21] DEBUG[23666] bridge_channel.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: 0xb66cad10(SIP/101-00000000) is joining [Oct 18 10:03:21] DEBUG[23666] bridge_channel.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: pushing 0xb66cad10(SIP/101-00000000) [Oct 18 10:03:21] DEBUG[23596] threadpool.c: Increasing threadpool stasis-core's size by 1 [Oct 18 10:03:21] VERBOSE[23666] bridge_channel.c: Channel SIP/101-00000000 joined 'simple_bridge' stasis-bridge [Oct 18 10:03:21] DEBUG[23666] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 18 10:03:21] DEBUG[23666] bridge_native_rtp.c: Bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5' can not use native RTP bridge as two channels are required [Oct 18 10:03:21] DEBUG[23666] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 18 10:03:21] DEBUG[23666] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 18 10:03:21] DEBUG[23666] bridge.c: Chose bridge technology simple_bridge [Oct 18 10:03:21] DEBUG[23666] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5 is already using the new technology. [Oct 18 10:03:21] DEBUG[23666] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: 0xb66cad10(SIP/101-00000000) is joining simple_bridge technology [Oct 18 10:03:21] DEBUG[23666] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 18 10:03:21] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 800): { "type": "ChannelEnteredBridge", "timestamp": "2016-10-18T10:03:21.887+0200", "bridge": { "id": "b1ecceb5-0432-40e6-b8c4-04dda2246cd5", "technology": "simple_bridge", "bridge_type": "mixing", "bridge_class": "stasis", "creator": "Stasis", "name": "", "channels": [ "atom_asterisk-1476777778.0" ] }, "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:21] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 800 [Oct 18 10:03:21] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 800 [Oct 18 10:03:21] DEBUG[23661] stasis/app.c: bridge '__AST_BRIDGE_ALL_TOPIC': is 0 interested in attendant [Oct 18 10:03:21] DEBUG[23661] stasis/app.c: bridge '__AST_BRIDGE_ALL_TOPIC' unsubscribed from attendant [Oct 18 10:03:21] DEBUG[23665] res_ari.c: Examining ARI response: 204 No Content [Oct 18 10:03:21] DEBUG[23661] stasis/app.c: Bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5' is 1 interested in attendant [Oct 18 10:03:21] DEBUG[23665] http.c: HTTP keeping session open. status_code:204 [Oct 18 10:03:21] DEBUG[23665] http.c: HTTP idle timeout or peer closed connection. [Oct 18 10:03:21] DEBUG[23665] http.c: HTTP closing session. Top level [Oct 18 10:03:24] DEBUG[23666] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 18 10:03:24] DEBUG[23666] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:24] DEBUG[23666] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:24] DEBUG[23666] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:24] DEBUG[23666] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:24] DEBUG[23666] acl.c: Attached to given IP address [Oct 18 10:03:24] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:24] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:24] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:24] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:24] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:28] DEBUG[23670] http.c: HTTP opening session. Top level [Oct 18 10:03:28] DEBUG[23670] http.c: HTTP Request URI is /ari/channels?endpoint=SIP/290&app=attendant [Oct 18 10:03:28] DEBUG[23670] http.c: match request [ari/channels] with handler [httpstatus] len 10 [Oct 18 10:03:28] DEBUG[23670] http.c: match request [ari/channels] with handler [amanager] len 8 [Oct 18 10:03:28] DEBUG[23670] http.c: match request [ari/channels] with handler [arawman] len 7 [Oct 18 10:03:28] DEBUG[23670] http.c: match request [ari/channels] with handler [manager] len 7 [Oct 18 10:03:28] DEBUG[23670] http.c: match request [ari/channels] with handler [rawman] len 6 [Oct 18 10:03:28] DEBUG[23670] http.c: match request [ari/channels] with handler [static] len 6 [Oct 18 10:03:28] DEBUG[23670] http.c: match request [ari/channels] with handler [amxml] len 5 [Oct 18 10:03:28] DEBUG[23670] http.c: match request [ari/channels] with handler [mxml] len 4 [Oct 18 10:03:28] DEBUG[23670] http.c: match request [ari/channels] with handler [ari] len 3 [Oct 18 10:03:28] DEBUG[23670] http.c: Match made with [ari] [Oct 18 10:03:28] DEBUG[23670] res_ari.c: Finding handler for channels [Oct 18 10:03:28] DEBUG[23670] res_ari.c: Checking events [Oct 18 10:03:28] DEBUG[23670] res_ari.c: Checking bridges [Oct 18 10:03:28] DEBUG[23670] res_ari.c: Checking channels [Oct 18 10:03:28] DEBUG[23670] res_ari.c: Got it! [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: Asked to create a SIP channel with formats: (slin) [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: Allocating new SIP dialog for 20c3799246c1c94d63e566ae45f4ae22@127.0.1.1:5060 - INVITE (No RTP) [Oct 18 10:03:28] DEBUG[23670] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x8f033c8' [Oct 18 10:03:28] DEBUG[23670] res_rtp_asterisk.c: Allocated port 6324 for RTP instance '0x8f033c8' [Oct 18 10:03:28] DEBUG[23670] res_rtp_asterisk.c: Creating ICE session 0.0.0.0:6324 (6324) for RTP instance '0x8f033c8' [Oct 18 10:03:28] DEBUG[23670] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:03:28] DEBUG[23670] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:03:28] DEBUG[23670] rtp_engine.c: RTP instance '0x8f033c8' is setup and ready to go [Oct 18 10:03:28] DEBUG[23670] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x8efbb68' [Oct 18 10:03:28] DEBUG[23670] res_rtp_asterisk.c: Allocated port 13654 for RTP instance '0x8efbb68' [Oct 18 10:03:28] DEBUG[23670] res_rtp_asterisk.c: Creating ICE session 0.0.0.0:13654 (13654) for RTP instance '0x8efbb68' [Oct 18 10:03:28] DEBUG[23670] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:03:28] DEBUG[23670] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:03:28] DEBUG[23670] rtp_engine.c: RTP instance '0x8efbb68' is setup and ready to go [Oct 18 10:03:28] DEBUG[23670] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x8efbb68' [Oct 18 10:03:28] DEBUG[23670] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x8f033c8' [Oct 18 10:03:28] VERBOSE[23670] netsock2.c: Using SIP RTP CoS mark 5 [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: Setting NAT on RTP to On [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: Setting NAT on TRTP to On [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Oct 18 10:03:28] DEBUG[23670] acl.c: For destination '192.168.210.111', our source address is '192.168.210.71'. [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.210.71:5060 [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: Setting NAT on RTP to On [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: Setting NAT on TRTP to On [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: SIP call-id changed from '20c3799246c1c94d63e566ae45f4ae22@127.0.1.1:5060' to '205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060' [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: *** Our native formats are (alaw) [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: *** Joint capabilities are (nothing) [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: *** Our capabilities are (alaw) [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: *** Our preferred formats from the incoming channel are (slin) [Oct 18 10:03:28] DEBUG[23670] chan_sip.c: This channel will not be able to handle video. [Oct 18 10:03:28] DEBUG[23670] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 18 10:03:28] DEBUG[23670] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 18 10:03:28] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 534): { "type": "ChannelCreated", "timestamp": "2016-10-18T10:03:28.345+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Down", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "s", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 534 [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 534 [Oct 18 10:03:28] DEBUG[23670] res_stasis.c: attendant: Subscribing to atom_asterisk-1476777808.1 [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Outgoing Call for 290 [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Updating call counter for outgoing call [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Call to peer '290' is 1 out of 2147483647 [Oct 18 10:03:28] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 290 [Oct 18 10:03:28] DEBUG[23670] res_ari.c: Examining ARI response: 200 OK Content-type: application/json { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Down", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" } [Oct 18 10:03:28] DEBUG[23607] chan_sip.c: Checking device state for peer 290 [Oct 18 10:03:28] DEBUG[23607] devicestate.c: Changing state for SIP/290 - state 6 (Ringing) [Oct 18 10:03:28] DEBUG[23670] http.c: HTTP keeping session open. status_code:200 [Oct 18 10:03:28] DEBUG[23635] app_queue.c: Device 'SIP/290' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Oct 18 10:03:28] DEBUG[23612] app_queue.c: Extension '290@LocalSets' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Oct 18 10:03:28] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 627): { "value": "205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060", "variable": "SIPCALLID", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:28.346+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Down", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "s", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 627 [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 627 [Oct 18 10:03:28] DEBUG[23670] http.c: HTTP idle timeout or peer closed connection. [Oct 18 10:03:28] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 606): { "type": "ChannelDialplan", "timestamp": "2016-10-18T10:03:28.347+0200", "dialplan_app": "AppDial2", "dialplan_app_data": "(Outgoing Line)", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Down", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 606 [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 606 [Oct 18 10:03:28] DEBUG[23670] http.c: HTTP closing session. Top level [Oct 18 10:03:28] DEBUG[23667] ari/ari_websockets.c: Examining ARI event (length 182): { "type": "DeviceStateChanged", "application": "attendant", "timestamp": "2016-10-18T10:03:28.350+0200", "device_state": { "name": "SIP/290", "state": "RINGING" } } [Oct 18 10:03:28] DEBUG[23667] res_http_websocket.c: Writing websocket string of length 182 [Oct 18 10:03:28] DEBUG[23667] res_http_websocket.c: Writing websocket text frame, length 182 [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: ** Our capability: (alaw) Video flag: False Text flag: False [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: ** Our prefcodec: (slin) [Oct 18 10:03:28] VERBOSE[23671] chan_sip.c: Audio is at 6324 [Oct 18 10:03:28] VERBOSE[23671] chan_sip.c: Adding codec alaw to SDP [Oct 18 10:03:28] VERBOSE[23671] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: -- Done with adding codecs to SDP [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Initializing initreq for method INVITE - callid 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 0 [ 43]: INVITE sip:290@192.168.210.111:5060 SIP/2.0 [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK7d3dc16b;rport [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 3 [ 66]: From: "Anonymous" ;tag=as24b2d10e [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 4 [ 34]: To: [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 5 [ 44]: Contact: [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 6 [ 61]: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 14.0.2 [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 9 [ 35]: Date: Tue, 18 Oct 2016 08:03:28 GMT [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Oct 18 10:03:28] VERBOSE[23671] chan_sip.c: Reliably Transmitting (NAT) to 192.168.210.111:5060: INVITE sip:290@192.168.210.111:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK7d3dc16b;rport Max-Forwards: 70 From: "Anonymous" ;tag=as24b2d10e To: Contact: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 14.0.2 Date: Tue, 18 Oct 2016 08:03:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 242 v=0 o=root 1155297451 1155297451 IN IP4 192.168.210.71 s=Asterisk PBX 14.0.2 c=IN IP4 192.168.210.71 t=0 0 m=audio 6324 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 --- [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Oct 18 10:03:28] DEBUG[23671] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.210.111:5060 [Oct 18 10:03:28] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 580): { "type": "Dial", "timestamp": "2016-10-18T10:03:28.352+0200", "dialstatus": "", "forward": "", "dialstring": "290", "peer": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Down", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 580 [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 580 [Oct 18 10:03:28] VERBOSE[23671] dial.c: Called 290 [Oct 18 10:03:28] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.111:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK7d3dc16b;rport=5060 From: "Anonymous" ;tag=as24b2d10e To: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 102 INVITE Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces, from-change Server: SIPPER for PhonerLite Content-Length: 0 <-------------> [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK7d3dc16b;rport=5060 [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as24b2d10e [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 3 [ 34]: To: [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 6 [ 78]: Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 7 [ 32]: Supported: replaces, from-change [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 8 [ 29]: Server: SIPPER for PhonerLite [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Oct 18 10:03:28] VERBOSE[23625] chan_sip.c: --- (10 headers 0 lines) --- [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: = Looking for Call ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 (Checking To) --From tag as24b2d10e --To-tag [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: *** SIP TIMER: Cancelling retransmission #15 - INVITE (got response) [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060' Request 102: Found [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: SIP response 100 to standard invite [Oct 18 10:03:28] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.111:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK7d3dc16b;rport=5060 From: "Anonymous" ;tag=as24b2d10e To: ;tag=80a63ee97693e611b2cfaca141d28d25 Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces, from-change Server: SIPPER for PhonerLite Content-Length: 0 <-------------> [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK7d3dc16b;rport=5060 [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as24b2d10e [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 3 [ 71]: To: ;tag=80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 6 [ 39]: Contact: [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 7 [ 78]: Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 8 [ 32]: Supported: replaces, from-change [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 9 [ 29]: Server: SIPPER for PhonerLite [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Oct 18 10:03:28] VERBOSE[23625] chan_sip.c: --- (11 headers 0 lines) --- [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: = Looking for Call ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 (Checking To) --From tag as24b2d10e --To-tag 80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060' Request 102: Found [Oct 18 10:03:28] DEBUG[23625] chan_sip.c: SIP response 180 to standard invite [Oct 18 10:03:28] VERBOSE[23625] sip/route.c: sip_route_dump: route/path hop: [Oct 18 10:03:28] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 290 [Oct 18 10:03:28] DEBUG[23607] chan_sip.c: Checking device state for peer 290 [Oct 18 10:03:28] VERBOSE[23671] dial.c: SIP/290-00000001 is ringing [Oct 18 10:03:28] DEBUG[23607] devicestate.c: Changing state for SIP/290 - state 6 (Ringing) [Oct 18 10:03:28] DEBUG[23596] threadpool.c: Increasing threadpool stasis-core's size by 1 [Oct 18 10:03:28] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 540): { "type": "ChannelStateChange", "timestamp": "2016-10-18T10:03:28.363+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Ringing", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 540 [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 540 [Oct 18 10:03:28] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 590): { "type": "Dial", "timestamp": "2016-10-18T10:03:28.364+0200", "dialstatus": "RINGING", "forward": "", "dialstring": "290", "peer": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Ringing", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 590 [Oct 18 10:03:28] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 590 [Oct 18 10:03:29] DEBUG[23666] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 18 10:03:29] DEBUG[23666] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:29] DEBUG[23666] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:29] DEBUG[23666] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:29] DEBUG[23666] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:29] DEBUG[23666] acl.c: Attached to given IP address [Oct 18 10:03:29] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:29] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:29] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:29] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:29] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:30] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.111:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK7d3dc16b;rport=5060 From: "Anonymous" ;tag=as24b2d10e To: ;tag=80a63ee97693e611b2cfaca141d28d25 Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces, from-change, gruu Server: SIPPER for PhonerLite Content-Length: 237 v=0 o=- 2923150685 1 IN IP4 192.168.210.111 s=SIPPER for PhonerLite c=IN IP4 192.168.210.111 t=0 0 m=audio 5062 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:2022430885 a=sendrecv <-------------> [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK7d3dc16b;rport=5060 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as24b2d10e [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 3 [ 71]: To: ;tag=80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 6 [ 39]: Contact: [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 8 [ 78]: Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 9 [ 38]: Supported: replaces, from-change, gruu [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 10 [ 29]: Server: SIPPER for PhonerLite [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 11 [ 19]: Content-Length: 237 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Header 12 [ 0]: [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Body 0 [ 3]: v=0 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Body 1 [ 39]: o=- 2923150685 1 IN IP4 192.168.210.111 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Body 2 [ 23]: s=SIPPER for PhonerLite [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Body 3 [ 24]: c=IN IP4 192.168.210.111 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Body 5 [ 26]: m=audio 5062 RTP/AVP 8 101 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Body 9 [ 17]: a=ssrc:2022430885 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Body 10 [ 10]: a=sendrecv [Oct 18 10:03:30] VERBOSE[23625] chan_sip.c: --- (12 headers 11 lines) --- [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: = Looking for Call ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 (Checking To) --From tag as24b2d10e --To-tag 80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Acked pending invite 102 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Stopping retransmission on '205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060' of Request 102: Match Found [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: SIP response 200 to standard invite [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Processing session-level SDP o=- 2923150685 1 IN IP4 192.168.210.111... OK. [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Processing session-level SDP s=SIPPER for PhonerLite... UNSUPPORTED OR FAILED. [Oct 18 10:03:30] DEBUG[23625] netsock2.c: Splitting '192.168.210.111' into... [Oct 18 10:03:30] DEBUG[23625] netsock2.c: ...host '192.168.210.111' and port ''. [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.210.111... OK. [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 18 10:03:30] VERBOSE[23625] chan_sip.c: Found RTP audio format 8 [Oct 18 10:03:30] DEBUG[23625] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb385047c [Oct 18 10:03:30] VERBOSE[23625] chan_sip.c: Found RTP audio format 101 [Oct 18 10:03:30] DEBUG[23625] rtp_engine.c: Setting tx payload type 101 based on m type on 0xb385047c [Oct 18 10:03:30] VERBOSE[23625] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 18 10:03:30] VERBOSE[23625] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=ssrc:2022430885... UNSUPPORTED OR FAILED. [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 18 10:03:30] VERBOSE[23625] chan_sip.c: Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 18 10:03:30] VERBOSE[23625] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 18 10:03:30] DEBUG[23625] res_rtp_asterisk.c: Set role to CONTROLLING (0x8f033c8) [Oct 18 10:03:30] DEBUG[23625] res_rtp_asterisk.c: Set role failed; no ice instance (0x8f033c8) [Oct 18 10:03:30] DEBUG[23625] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8f033c8' [Oct 18 10:03:30] VERBOSE[23625] chan_sip.c: Peer audio RTP is at port 192.168.210.111:5062 [Oct 18 10:03:30] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 8 (0xb6402fd0) from 0xb385047c to 0x8f03574 [Oct 18 10:03:30] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 101 (0x8bedf80) from 0xb385047c to 0x8f03574 [Oct 18 10:03:30] DEBUG[23625] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x8f033c8' [Oct 18 10:03:30] DEBUG[23625] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8efbb68' [Oct 18 10:03:30] VERBOSE[23625] chan_sip.c: Peer doesn't provide T.140 [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: We're settling with these formats: (alaw) [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: We have an owner, now see if we need to change this call [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Updating call counter for outgoing call [Oct 18 10:03:30] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 290 [Oct 18 10:03:30] DEBUG[23607] chan_sip.c: Checking device state for peer 290 [Oct 18 10:03:30] DEBUG[23607] devicestate.c: Changing state for SIP/290 - state 2 (In use) [Oct 18 10:03:30] VERBOSE[23625] sip/route.c: sip_route_dump: route/path hop: [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Strict routing enforced for session 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:03:30] VERBOSE[23625] chan_sip.c: Transmitting (NAT) to 192.168.210.111:5060: ACK sip:290@192.168.210.111:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK3640df60;rport Max-Forwards: 70 From: "Anonymous" ;tag=as24b2d10e To: ;tag=80a63ee97693e611b2cfaca141d28d25 Contact: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 14.0.2 Content-Length: 0 --- [Oct 18 10:03:30] DEBUG[23625] chan_sip.c: Trying to put 'ACK sip:290' onto UDP socket destined for 192.168.210.111:5060 [Oct 18 10:03:30] DEBUG[23635] app_queue.c: Device 'SIP/290' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 18 10:03:30] DEBUG[23667] ari/ari_websockets.c: Examining ARI event (length 180): { "type": "DeviceStateChanged", "application": "attendant", "timestamp": "2016-10-18T10:03:30.462+0200", "device_state": { "name": "SIP/290", "state": "INUSE" } } [Oct 18 10:03:30] DEBUG[23667] res_http_websocket.c: Writing websocket string of length 180 [Oct 18 10:03:30] DEBUG[23667] res_http_websocket.c: Writing websocket text frame, length 180 [Oct 18 10:03:30] DEBUG[23612] app_queue.c: Extension '290@LocalSets' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 18 10:03:30] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 535): { "type": "ChannelStateChange", "timestamp": "2016-10-18T10:03:30.462+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:30] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 535 [Oct 18 10:03:30] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 535 [Oct 18 10:03:30] VERBOSE[23671] dial.c: SIP/290-00000001 answered [Oct 18 10:03:30] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 290 [Oct 18 10:03:30] DEBUG[23607] chan_sip.c: Checking device state for peer 290 [Oct 18 10:03:30] DEBUG[23607] devicestate.c: Changing state for SIP/290 - state 2 (In use) [Oct 18 10:03:30] VERBOSE[23671] ari/resource_channels.c: Launching Stasis(attendant) on SIP/290-00000001 [Oct 18 10:03:30] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 584): { "type": "Dial", "timestamp": "2016-10-18T10:03:30.464+0200", "dialstatus": "ANSWER", "forward": "", "dialstring": "290", "peer": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:30] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 584 [Oct 18 10:03:30] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 584 [Oct 18 10:03:30] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 575): { "value": "", "variable": "STASISSTATUS", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:30.465+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:30] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 575 [Oct 18 10:03:30] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 575 [Oct 18 10:03:30] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 542): { "type": "StasisStart", "timestamp": "2016-10-18T10:03:30.465+0200", "args": [], "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:30] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 542 [Oct 18 10:03:30] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 542 [Oct 18 10:03:30] DEBUG[23671] res_rtp_asterisk.c: Got RTCP report of 60 bytes [Oct 18 10:03:30] DEBUG[23671] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:30] DEBUG[23671] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:30] DEBUG[23671] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:30] DEBUG[23671] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:30] DEBUG[23671] acl.c: Attached to given IP address [Oct 18 10:03:30] DEBUG[23671] res_rtp_asterisk.c: 0x8f03e60 -- Probation learning mode pass with source address 192.168.210.111:5062 [Oct 18 10:03:30] VERBOSE[23671] res_rtp_asterisk.c: 0x8f03e60 -- Probation passed - setting RTP source address to 192.168.210.111:5062 [Oct 18 10:03:34] DEBUG[23666] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 18 10:03:34] DEBUG[23666] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:34] DEBUG[23666] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:34] DEBUG[23666] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:34] DEBUG[23666] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:34] DEBUG[23666] acl.c: Attached to given IP address [Oct 18 10:03:34] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:34] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:34] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:34] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:34] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:35] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:35] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:35] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:35] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:35] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:39] DEBUG[23666] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 18 10:03:39] DEBUG[23666] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:39] DEBUG[23666] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:39] DEBUG[23666] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:39] DEBUG[23666] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:39] DEBUG[23666] acl.c: Attached to given IP address [Oct 18 10:03:39] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:39] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:39] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:39] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:39] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:39] DEBUG[23671] res_rtp_asterisk.c: Got RTCP report of 60 bytes [Oct 18 10:03:39] DEBUG[23671] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:39] DEBUG[23671] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:39] DEBUG[23671] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:39] DEBUG[23671] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:39] DEBUG[23671] acl.c: Attached to given IP address [Oct 18 10:03:40] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:40] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:40] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:40] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:40] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:03:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:03:44] DEBUG[23666] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 18 10:03:44] DEBUG[23666] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:44] DEBUG[23666] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:44] DEBUG[23666] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:44] DEBUG[23666] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:44] DEBUG[23666] acl.c: Attached to given IP address [Oct 18 10:03:44] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:44] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:44] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:44] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:44] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:45] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:45] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:03:45] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:03:45] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:03:45] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:03:46] DEBUG[23675] http.c: HTTP opening session. Top level [Oct 18 10:03:46] DEBUG[23675] http.c: HTTP Request URI is /ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel?channel=atom_asterisk-1476777808.1 [Oct 18 10:03:46] DEBUG[23675] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [httpstatus] len 10 [Oct 18 10:03:46] DEBUG[23675] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [amanager] len 8 [Oct 18 10:03:46] DEBUG[23675] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [arawman] len 7 [Oct 18 10:03:46] DEBUG[23675] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [manager] len 7 [Oct 18 10:03:46] DEBUG[23675] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [rawman] len 6 [Oct 18 10:03:46] DEBUG[23675] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [static] len 6 [Oct 18 10:03:46] DEBUG[23675] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [amxml] len 5 [Oct 18 10:03:46] DEBUG[23675] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [mxml] len 4 [Oct 18 10:03:46] DEBUG[23675] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5/addChannel] with handler [ari] len 3 [Oct 18 10:03:46] DEBUG[23675] http.c: Match made with [ari] [Oct 18 10:03:46] DEBUG[23675] res_ari.c: Finding handler for bridges [Oct 18 10:03:46] DEBUG[23675] res_ari.c: Checking events [Oct 18 10:03:46] DEBUG[23675] res_ari.c: Checking bridges [Oct 18 10:03:46] DEBUG[23675] res_ari.c: Got it! [Oct 18 10:03:46] DEBUG[23675] res_ari.c: Finding handler for b1ecceb5-0432-40e6-b8c4-04dda2246cd5 [Oct 18 10:03:46] DEBUG[23675] res_ari.c: Checking bridgeId [Oct 18 10:03:46] DEBUG[23675] res_ari.c: Got it! [Oct 18 10:03:46] DEBUG[23675] res_ari.c: Finding handler for addChannel [Oct 18 10:03:46] DEBUG[23675] res_ari.c: Checking addChannel [Oct 18 10:03:46] DEBUG[23675] res_ari.c: Got it! [Oct 18 10:03:46] DEBUG[23675] stasis/control.c: atom_asterisk-1476777808.1: Sending channel add_to_bridge command [Oct 18 10:03:46] DEBUG[23671] bridge_roles.c: Roles did not exist on channel SIP/290-00000001 [Oct 18 10:03:46] DEBUG[23671] stasis/control.c: atom_asterisk-1476777808.1: Adding to bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5 [Oct 18 10:03:46] DEBUG[23671] stasis/app.c: Bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5' is 2 interested in attendant [Oct 18 10:03:46] DEBUG[23676] bridge_channel.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: 0xb663ca88(SIP/290-00000001) is joining [Oct 18 10:03:46] DEBUG[23676] bridge_channel.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: pushing 0xb663ca88(SIP/290-00000001) [Oct 18 10:03:46] VERBOSE[23676] bridge_channel.c: Channel SIP/290-00000001 joined 'simple_bridge' stasis-bridge [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 18 10:03:46] DEBUG[23676] bridge.c: Chose bridge technology native_rtp [Oct 18 10:03:46] VERBOSE[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: switching from simple_bridge technology to native_rtp [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling native_rtp technology constructor [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: moving 0xb66cad10(SIP/101-00000000) to dummy bridge temporarily [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: 0xb66cad10(SIP/101-00000000) is leaving simple_bridge technology (dummy) [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling simple_bridge technology stop [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: 0xb663ca88(SIP/290-00000001) is joining native_rtp technology [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Sending reinvite on SIP '205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060' - It's audio soon redirected to IP 192.168.210.40:11788 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Strict routing enforced for session 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 836): { "type": "ChannelEnteredBridge", "timestamp": "2016-10-18T10:03:46.015+0200", "bridge": { "id": "b1ecceb5-0432-40e6-b8c4-04dda2246cd5", "technology": "simple_bridge", "bridge_type": "mixing", "bridge_class": "stasis", "creator": "Stasis", "name": "", "channels": [ "atom_asterisk-1476777778.0", "atom_asterisk-1476777808.1" ] }, "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 836 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 836 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: ** Our native-bridge filtered capablity: (alaw) [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: ** Our prefcodec: (slin) [Oct 18 10:03:46] VERBOSE[23676] chan_sip.c: Audio is at 6324 [Oct 18 10:03:46] VERBOSE[23676] chan_sip.c: Adding codec alaw to SDP [Oct 18 10:03:46] VERBOSE[23676] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: -- Done with adding codecs to SDP [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Initializing already initialized SIP dialog 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 (presumably reinvite) [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 0 [ 43]: INVITE sip:290@192.168.210.111:5060 SIP/2.0 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK1cf3fe94;rport [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 3 [ 66]: From: "Anonymous" ;tag=as24b2d10e [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 4 [ 71]: To: ;tag=80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 5 [ 44]: Contact: [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 6 [ 61]: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 14.0.2 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 9 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Oct 18 10:03:46] VERBOSE[23676] chan_sip.c: Reliably Transmitting (NAT) to 192.168.210.111:5060: INVITE sip:290@192.168.210.111:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK1cf3fe94;rport Max-Forwards: 70 From: "Anonymous" ;tag=as24b2d10e To: ;tag=80a63ee97693e611b2cfaca141d28d25 Contact: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 14.0.2 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: 243 v=0 o=root 1155297451 1155297452 IN IP4 192.168.210.40 s=Asterisk PBX 14.0.2 c=IN IP4 192.168.210.40 t=0 0 m=audio 11788 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 --- [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #27 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.210.111:5060 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Sending reinvite on SIP '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' - It's audio soon redirected to IP 192.168.210.111:5062 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Strict routing enforced for session 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: ** Our native-bridge filtered capablity: (alaw) [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: ** Our prefcodec: (slin) [Oct 18 10:03:46] VERBOSE[23676] chan_sip.c: Audio is at 8570 [Oct 18 10:03:46] VERBOSE[23676] chan_sip.c: Adding codec alaw to SDP [Oct 18 10:03:46] VERBOSE[23676] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: -- Done with adding codecs to SDP [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Initializing already initialized SIP dialog 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 (presumably reinvite) [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 0 [ 42]: INVITE sip:101@192.168.210.40:5062 SIP/2.0 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK06a1525e;rport [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 3 [ 66]: From: "Anonymous" ;tag=as3af1af94 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 4 [ 47]: To: ;tag=673443222 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 5 [ 44]: Contact: [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 6 [ 61]: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 14.0.2 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 9 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.111:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK1cf3fe94;rport=5060 From: "Anonymous" ;tag=as24b2d10e To: ;tag=80a63ee97693e611b2cfaca141d28d25 Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 103 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces, from-change Server: SIPPER for PhonerLite Content-Length: 237 v=0 o=- 2923150685 2 IN IP4 192.168.210.111 s=SIPPER for PhonerLite c=IN IP4 192.168.210.111 t=0 0 m=audio 5062 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:2022430885 a=sendrecv <-------------> [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 18 10:03:46] VERBOSE[23676] chan_sip.c: Reliably Transmitting (NAT) to 192.168.210.40:5062: INVITE sip:101@192.168.210.40:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK06a1525e;rport Max-Forwards: 70 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Contact: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 14.0.2 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 1361751049 1361751050 IN IP4 192.168.210.111 s=Asterisk PBX 14.0.2 c=IN IP4 192.168.210.111 t=0 0 m=audio 5062 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 --- [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK1cf3fe94;rport=5060 [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #28 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as24b2d10e [Oct 18 10:03:46] DEBUG[23676] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.210.40:5062 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 3 [ 71]: To: ;tag=80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:03:46] VERBOSE[23676] bridge_native_rtp.c: Remotely bridged 'SIP/290-00000001' and 'SIP/101-00000000' - media will flow directly between them [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 6 [ 39]: Contact: [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: 0xb66cad10(SIP/101-00000000) is joining native_rtp technology [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 8 [ 78]: Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 9 [ 32]: Supported: replaces, from-change [Oct 18 10:03:46] VERBOSE[23676] bridge_native_rtp.c: Remotely bridged 'SIP/290-00000001' and 'SIP/101-00000000' - media will flow directly between them [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 10 [ 29]: Server: SIPPER for PhonerLite [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling native_rtp technology start [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 11 [ 19]: Content-Length: 237 [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling simple_bridge technology destructor [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 12 [ 0]: [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 0 [ 3]: v=0 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 1 [ 39]: o=- 2923150685 2 IN IP4 192.168.210.111 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 2 [ 23]: s=SIPPER for PhonerLite [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 3 [ 24]: c=IN IP4 192.168.210.111 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 5 [ 26]: m=audio 5062 RTP/AVP 8 101 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 9 [ 17]: a=ssrc:2022430885 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 10 [ 10]: a=sendrecv [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: --- (12 headers 11 lines) --- [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: = Looking for Call ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 (Checking To) --From tag as24b2d10e --To-tag 80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Acked pending invite 103 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #27 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Stopping retransmission on '205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060' of Request 103: Match Found [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: SIP response 200 to RE-invite on outgoing call 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing session-level SDP o=- 2923150685 2 IN IP4 192.168.210.111... OK. [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing session-level SDP s=SIPPER for PhonerLite... UNSUPPORTED OR FAILED. [Oct 18 10:03:46] DEBUG[23625] netsock2.c: Splitting '192.168.210.111' into... [Oct 18 10:03:46] DEBUG[23625] netsock2.c: ...host '192.168.210.111' and port ''. [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.210.111... OK. [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Found RTP audio format 8 [Oct 18 10:03:46] DEBUG[23625] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb385047c [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Found RTP audio format 101 [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 589): { "value": "SIP/101-00000000", "variable": "BRIDGEPEER", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.020+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23625] rtp_engine.c: Setting tx payload type 101 based on m type on 0xb385047c [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 589 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 589 [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=ssrc:2022430885... UNSUPPORTED OR FAILED. [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 18 10:03:46] DEBUG[23625] res_rtp_asterisk.c: Set role to CONTROLLING (0x8f033c8) [Oct 18 10:03:46] DEBUG[23625] res_rtp_asterisk.c: Set role failed; no ice instance (0x8f033c8) [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Peer audio RTP is at port 192.168.210.111:5062 [Oct 18 10:03:46] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 8 (0xb6406f48) from 0xb385047c to 0x8f03574 [Oct 18 10:03:46] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 101 (0x8bedf80) from 0xb385047c to 0x8f03574 [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 630): { "value": "1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060", "variable": "BRIDGEPVTCALLID", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.020+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 630 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 630 [Oct 18 10:03:46] DEBUG[23625] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x8f033c8' [Oct 18 10:03:46] DEBUG[23625] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8efbb68' [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Peer doesn't provide T.140 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: We're settling with these formats: (alaw) [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: We have an owner, now see if we need to change this call [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Updating call counter for outgoing call [Oct 18 10:03:46] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 290 [Oct 18 10:03:46] DEBUG[23607] chan_sip.c: Checking device state for peer 290 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Strict routing enforced for session 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:03:46] DEBUG[23607] devicestate.c: Changing state for SIP/290 - state 2 (In use) [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 589): { "value": "SIP/290-00000001", "variable": "BRIDGEPEER", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.020+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 589 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 589 [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Transmitting (NAT) to 192.168.210.111:5060: ACK sip:290@192.168.210.111:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK213a4ef0;rport Max-Forwards: 70 From: "Anonymous" ;tag=as24b2d10e To: ;tag=80a63ee97693e611b2cfaca141d28d25 Contact: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 14.0.2 Content-Length: 0 --- [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Trying to put 'ACK sip:290' onto UDP socket destined for 192.168.210.111:5060 [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 630): { "value": "205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060", "variable": "BRIDGEPVTCALLID", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.020+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 630 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 630 [Oct 18 10:03:46] DEBUG[23676] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 18 10:03:46] DEBUG[23671] stasis/app.c: App 'attendant' not subscribed to bridge '__AST_BRIDGE_ALL_TOPIC' [Oct 18 10:03:46] DEBUG[23671] stasis/app.c: Bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5' is 3 interested in attendant [Oct 18 10:03:46] DEBUG[23675] res_ari.c: Examining ARI response: 204 No Content [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 18 10:03:46] DEBUG[23675] http.c: HTTP keeping session open. status_code:204 [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 18 10:03:46] DEBUG[23676] bridge.c: Chose bridge technology native_rtp [Oct 18 10:03:46] DEBUG[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5 is already using the new technology. [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK06a1525e;rport=5060 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 103 INVITE User-Agent: Yealink SIP-T21P_E2 52.80.0.95 Content-Length: 0 <-------------> [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK06a1525e;rport=5060 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as3af1af94 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 3 [ 47]: To: ;tag=673443222 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 6 [ 42]: User-Agent: Yealink SIP-T21P_E2 52.80.0.95 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: --- (8 headers 0 lines) --- [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: = Looking for Call ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 (Checking To) --From tag as3af1af94 --To-tag 673443222 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: *** SIP TIMER: Cancelling retransmission #28 - INVITE (got response) [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' Request 103: Found [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: SIP response 100 to RE-invite on outgoing call 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:03:46] DEBUG[23666] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 18 10:03:46] DEBUG[23666] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 18 10:03:46] DEBUG[23666] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 589): { "value": "SIP/101-00000000", "variable": "BRIDGEPEER", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.025+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23666] bridge.c: Chose bridge technology native_rtp [Oct 18 10:03:46] DEBUG[23666] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5 is already using the new technology. [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 589 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 589 [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 630): { "value": "1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060", "variable": "BRIDGEPVTCALLID", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.027+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 630 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 630 [Oct 18 10:03:46] DEBUG[23675] http.c: HTTP idle timeout or peer closed connection. [Oct 18 10:03:46] DEBUG[23675] http.c: HTTP closing session. Top level [Oct 18 10:03:46] DEBUG[23676] res_rtp_asterisk.c: Ooh, format changed from none to alaw [Oct 18 10:03:46] DEBUG[23676] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x8f033c8' [Oct 18 10:03:46] DEBUG[23676] res_rtp_asterisk.c: 0x8f03e60 -- Probation learning mode pass with source address 192.168.210.111:5062 [Oct 18 10:03:46] VERBOSE[23676] res_rtp_asterisk.c: 0x8f03e60 -- Probation passed - setting RTP source address to 192.168.210.111:5062 [Oct 18 10:03:46] DEBUG[23666] res_rtp_asterisk.c: Ooh, format changed from none to alaw [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 589): { "value": "SIP/290-00000001", "variable": "BRIDGEPEER", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.027+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 589 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 589 [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 630): { "value": "205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060", "variable": "BRIDGEPVTCALLID", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.027+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 630 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 630 [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 589): { "value": "SIP/101-00000000", "variable": "BRIDGEPEER", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.028+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 589 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 589 [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK06a1525e;rport=5060 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 103 INVITE Contact: Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE User-Agent: Yealink SIP-T21P_E2 52.80.0.95 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 215 v=0 o=- 20002 20003 IN IP4 192.168.210.40 s=SDP data c=IN IP4 192.168.210.40 t=0 0 m=audio 11788 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK06a1525e;rport=5060 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as3af1af94 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 3 [ 47]: To: ;tag=673443222 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 6 [ 38]: Contact: [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 8 [115]: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 9 [ 42]: User-Agent: Yealink SIP-T21P_E2 52.80.0.95 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 10 [ 51]: Allow-Events: talk,hold,conference,refer,check-sync [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 11 [ 19]: Content-Length: 215 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Header 12 [ 0]: [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 0 [ 3]: v=0 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 1 [ 37]: o=- 20002 20003 IN IP4 192.168.210.40 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 2 [ 10]: s=SDP data [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.210.40 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 5 [ 27]: m=audio 11788 RTP/AVP 8 101 [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 630): { "value": "1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060", "variable": "BRIDGEPVTCALLID", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.028+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 630 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 7 [ 10]: a=ptime:20 [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 630 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Body 10 [ 10]: a=sendrecv [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: --- (12 headers 11 lines) --- [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: = Looking for Call ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 (Checking To) --From tag as3af1af94 --To-tag 673443222 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Acked pending invite 103 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Stopping retransmission on '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' of Request 103: Match Found [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: SIP response 200 to RE-invite on outgoing call 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 589): { "value": "SIP/290-00000001", "variable": "BRIDGEPEER", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.029+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing session-level SDP o=- 20002 20003 IN IP4 192.168.210.40... OK. [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 589 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing session-level SDP s=SDP data... UNSUPPORTED OR FAILED. [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 589 [Oct 18 10:03:46] DEBUG[23625] netsock2.c: Splitting '192.168.210.40' into... [Oct 18 10:03:46] DEBUG[23625] netsock2.c: ...host '192.168.210.40' and port ''. [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.210.40... OK. [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Found RTP audio format 8 [Oct 18 10:03:46] DEBUG[23625] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb385047c [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Found RTP audio format 101 [Oct 18 10:03:46] DEBUG[23625] rtp_engine.c: Setting tx payload type 101 based on m type on 0xb385047c [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 18 10:03:46] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 630): { "value": "205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060", "variable": "BRIDGEPVTCALLID", "type": "ChannelVarset", "timestamp": "2016-10-18T10:03:46.029+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 630 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 18 10:03:46] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 630 [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 18 10:03:46] DEBUG[23625] res_rtp_asterisk.c: Set role to CONTROLLING (0xb663a450) [Oct 18 10:03:46] DEBUG[23625] res_rtp_asterisk.c: Set role failed; no ice instance (0xb663a450) [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Peer audio RTP is at port 192.168.210.40:11788 [Oct 18 10:03:46] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 8 (0xb64041f8) from 0xb385047c to 0xb663a5fc [Oct 18 10:03:46] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 101 (0x8bedf80) from 0xb385047c to 0xb663a5fc [Oct 18 10:03:46] DEBUG[23625] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb663a450' [Oct 18 10:03:46] DEBUG[23625] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb666b8d0' [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Peer doesn't provide T.140 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: We're settling with these formats: (alaw) [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: We have an owner, now see if we need to change this call [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Updating call counter for outgoing call [Oct 18 10:03:46] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 101 [Oct 18 10:03:46] DEBUG[23607] chan_sip.c: Checking device state for peer 101 [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Strict routing enforced for session 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:03:46] DEBUG[23607] devicestate.c: Changing state for SIP/101 - state 2 (In use) [Oct 18 10:03:46] VERBOSE[23625] chan_sip.c: Transmitting (NAT) to 192.168.210.40:5062: ACK sip:101@192.168.210.40:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK055f3bc4;rport Max-Forwards: 70 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Contact: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 14.0.2 Content-Length: 0 --- [Oct 18 10:03:46] DEBUG[23625] chan_sip.c: Trying to put 'ACK sip:101' onto UDP socket destined for 192.168.210.40:5062 [Oct 18 10:03:46] DEBUG[23666] res_rtp_asterisk.c: 0xb665a2c0 -- Probation learning mode pass with source address 192.168.210.40:11788 [Oct 18 10:03:46] VERBOSE[23666] res_rtp_asterisk.c: 0xb665a2c0 -- Probation passed - setting RTP source address to 192.168.210.40:11788 [Oct 18 10:04:03] DEBUG[23679] http.c: HTTP opening session. Top level [Oct 18 10:04:03] DEBUG[23679] http.c: HTTP Request URI is /ari/channels/atom_asterisk-1476777778.0 [Oct 18 10:04:03] DEBUG[23679] http.c: match request [ari/channels/atom_asterisk-1476777778.0] with handler [httpstatus] len 10 [Oct 18 10:04:03] DEBUG[23679] http.c: match request [ari/channels/atom_asterisk-1476777778.0] with handler [amanager] len 8 [Oct 18 10:04:03] DEBUG[23679] http.c: match request [ari/channels/atom_asterisk-1476777778.0] with handler [arawman] len 7 [Oct 18 10:04:03] DEBUG[23679] http.c: match request [ari/channels/atom_asterisk-1476777778.0] with handler [manager] len 7 [Oct 18 10:04:03] DEBUG[23679] http.c: match request [ari/channels/atom_asterisk-1476777778.0] with handler [rawman] len 6 [Oct 18 10:04:03] DEBUG[23679] http.c: match request [ari/channels/atom_asterisk-1476777778.0] with handler [static] len 6 [Oct 18 10:04:03] DEBUG[23679] http.c: match request [ari/channels/atom_asterisk-1476777778.0] with handler [amxml] len 5 [Oct 18 10:04:03] DEBUG[23679] http.c: match request [ari/channels/atom_asterisk-1476777778.0] with handler [mxml] len 4 [Oct 18 10:04:03] DEBUG[23679] http.c: match request [ari/channels/atom_asterisk-1476777778.0] with handler [ari] len 3 [Oct 18 10:04:03] DEBUG[23679] http.c: Match made with [ari] [Oct 18 10:04:03] DEBUG[23679] res_ari.c: Finding handler for channels [Oct 18 10:04:03] DEBUG[23679] res_ari.c: Checking events [Oct 18 10:04:03] DEBUG[23679] res_ari.c: Checking bridges [Oct 18 10:04:03] DEBUG[23679] res_ari.c: Checking channels [Oct 18 10:04:03] DEBUG[23679] res_ari.c: Got it! [Oct 18 10:04:03] DEBUG[23679] res_ari.c: Finding handler for atom_asterisk-1476777778.0 [Oct 18 10:04:03] DEBUG[23679] res_ari.c: Checking create [Oct 18 10:04:03] DEBUG[23679] res_ari.c: Checking channelId [Oct 18 10:04:03] DEBUG[23679] res_ari.c: Got it! [Oct 18 10:04:03] DEBUG[23679] channel.c: Soft-Hanging (0x20) up channel 'SIP/101-00000000' [Oct 18 10:04:03] DEBUG[23679] res_ari.c: Examining ARI response: 204 No Content [Oct 18 10:04:03] DEBUG[23679] http.c: HTTP keeping session open. status_code:204 [Oct 18 10:04:03] DEBUG[23666] bridge_channel.c: Setting 0xb66cad10(SIP/101-00000000) state from:0 to:1 [Oct 18 10:04:03] DEBUG[23666] bridge_channel.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: pulling 0xb66cad10(SIP/101-00000000) [Oct 18 10:04:03] VERBOSE[23666] bridge_channel.c: Channel SIP/101-00000000 left 'native_rtp' stasis-bridge [Oct 18 10:04:03] DEBUG[23666] bridge_channel.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: 0xb66cad10(SIP/101-00000000) is leaving native_rtp technology [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Sending reinvite on SIP '205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060' - It's audio soon redirected to IP 192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23666] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x8f033c8' [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Strict routing enforced for session 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: ** Our prefcodec: (slin) [Oct 18 10:04:03] VERBOSE[23666] chan_sip.c: Audio is at 6324 [Oct 18 10:04:03] VERBOSE[23666] chan_sip.c: Adding codec alaw to SDP [Oct 18 10:04:03] VERBOSE[23666] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: -- Done with adding codecs to SDP [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Initializing already initialized SIP dialog 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 (presumably reinvite) [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 0 [ 43]: INVITE sip:290@192.168.210.111:5060 SIP/2.0 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK1080baed;rport [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 3 [ 66]: From: "Anonymous" ;tag=as24b2d10e [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 4 [ 71]: To: ;tag=80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 5 [ 44]: Contact: [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 6 [ 61]: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 14.0.2 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 9 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Oct 18 10:04:03] DEBUG[23679] http.c: HTTP idle timeout or peer closed connection. [Oct 18 10:04:03] VERBOSE[23666] chan_sip.c: Reliably Transmitting (NAT) to 192.168.210.111:5060: INVITE sip:290@192.168.210.111:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK1080baed;rport Max-Forwards: 70 From: "Anonymous" ;tag=as24b2d10e To: ;tag=80a63ee97693e611b2cfaca141d28d25 Contact: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 14.0.2 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: 242 v=0 o=root 1155297451 1155297453 IN IP4 192.168.210.71 s=Asterisk PBX 14.0.2 c=IN IP4 192.168.210.71 t=0 0 m=audio 6324 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 --- [Oct 18 10:04:03] DEBUG[23679] http.c: HTTP closing session. Top level [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #12 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.210.111:5060 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Sending reinvite on SIP '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' - It's audio soon redirected to IP 192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23666] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb663a450' [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Strict routing enforced for session 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: ** Our prefcodec: (slin) [Oct 18 10:04:03] VERBOSE[23666] chan_sip.c: Audio is at 8570 [Oct 18 10:04:03] VERBOSE[23666] chan_sip.c: Adding codec alaw to SDP [Oct 18 10:04:03] VERBOSE[23666] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: -- Done with adding codecs to SDP [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Initializing already initialized SIP dialog 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 (presumably reinvite) [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 0 [ 42]: INVITE sip:101@192.168.210.40:5062 SIP/2.0 [Oct 18 10:04:03] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 568): { "cause": 32, "soft": true, "type": "ChannelHangupRequest", "timestamp": "2016-10-18T10:04:03.037+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK2603fe63;rport [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 568 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 568 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 3 [ 66]: From: "Anonymous" ;tag=as3af1af94 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 4 [ 47]: To: ;tag=673443222 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 5 [ 44]: Contact: [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 6 [ 61]: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 14.0.2 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 9 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Oct 18 10:04:03] VERBOSE[23666] chan_sip.c: Reliably Transmitting (NAT) to 192.168.210.40:5062: INVITE sip:101@192.168.210.40:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK2603fe63;rport Max-Forwards: 70 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Contact: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 14.0.2 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: 242 v=0 o=root 1361751049 1361751051 IN IP4 192.168.210.71 s=Asterisk PBX 14.0.2 c=IN IP4 192.168.210.71 t=0 0 m=audio 8570 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 --- [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #13 [Oct 18 10:04:03] DEBUG[23666] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.210.40:5062 [Oct 18 10:04:03] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 573): { "value": "", "variable": "BRIDGEPEER", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:03.037+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 573 [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 573 [Oct 18 10:04:03] DEBUG[23666] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/290-00000001' and 'SIP/101-00000000' - media will flow through Asterisk core [Oct 18 10:04:03] DEBUG[23666] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 18 10:04:03] DEBUG[23666] bridge_native_rtp.c: Bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5' can not use native RTP bridge as two channels are required [Oct 18 10:04:03] DEBUG[23666] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 18 10:04:03] DEBUG[23666] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 18 10:04:03] DEBUG[23666] bridge.c: Chose bridge technology simple_bridge [Oct 18 10:04:03] VERBOSE[23666] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: switching from native_rtp technology to simple_bridge [Oct 18 10:04:03] DEBUG[23666] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling simple_bridge technology constructor [Oct 18 10:04:03] DEBUG[23666] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: moving 0xb663ca88(SIP/290-00000001) to dummy bridge temporarily [Oct 18 10:04:03] DEBUG[23666] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: 0xb663ca88(SIP/290-00000001) is leaving native_rtp technology (dummy) [Oct 18 10:04:03] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 578): { "value": "", "variable": "BRIDGEPVTCALLID", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:03.037+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 578 [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 578 [Oct 18 10:04:03] DEBUG[23666] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling native_rtp technology stop [Oct 18 10:04:03] DEBUG[23666] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: 0xb663ca88(SIP/290-00000001) is joining simple_bridge technology [Oct 18 10:04:03] DEBUG[23666] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling simple_bridge technology start [Oct 18 10:04:03] DEBUG[23666] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling native_rtp technology destructor [Oct 18 10:04:03] DEBUG[23666] stasis/control.c: atom_asterisk-1476777778.0, b1ecceb5-0432-40e6-b8c4-04dda2246cd5: Channel leaving bridge [Oct 18 10:04:03] DEBUG[23666] stasis/app.c: bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5': is 2 interested in attendant [Oct 18 10:04:03] DEBUG[23661] stasis/control.c: atom_asterisk-1476777778.0: Channel departing bridge [Oct 18 10:04:03] DEBUG[23676] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 18 10:04:03] DEBUG[23676] bridge_native_rtp.c: Bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5' can not use native RTP bridge as two channels are required [Oct 18 10:04:03] DEBUG[23661] bridge.c: Waiting for 0xb66cad10(SIP/101-00000000) bridge thread to die. [Oct 18 10:04:03] DEBUG[23676] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 18 10:04:03] DEBUG[23676] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 18 10:04:03] DEBUG[23676] bridge.c: Chose bridge technology simple_bridge [Oct 18 10:04:03] DEBUG[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5 is already using the new technology. [Oct 18 10:04:03] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 794): { "type": "ChannelLeftBridge", "timestamp": "2016-10-18T10:04:03.041+0200", "bridge": { "id": "b1ecceb5-0432-40e6-b8c4-04dda2246cd5", "technology": "native_rtp", "bridge_type": "mixing", "bridge_class": "stasis", "creator": "Stasis", "name": "", "channels": [ "atom_asterisk-1476777808.1" ] }, "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 794 [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 794 [Oct 18 10:04:03] DEBUG[23666] stasis/control.c: reason: Channel was departed from bridge [Oct 18 10:04:03] DEBUG[23661] stasis/app.c: App 'attendant' not subscribed to bridge '__AST_BRIDGE_ALL_TOPIC' [Oct 18 10:04:03] DEBUG[23661] stasis/app.c: App 'attendant' not subscribed to channel 'atom_asterisk-1476777778.0' [Oct 18 10:04:03] DEBUG[23661] channel.c: Hanging up channel 'SIP/101-00000000' [Oct 18 10:04:03] DEBUG[23661] chan_sip.c: Hangup call SIP/101-00000000, SIP callid 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23661] chan_sip.c: update_call_counter(101) - decrement call limit counter on hangup [Oct 18 10:04:03] DEBUG[23661] chan_sip.c: Updating call counter for outgoing call [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.111:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK1080baed;rport=5060 From: "Anonymous" ;tag=as24b2d10e To: ;tag=80a63ee97693e611b2cfaca141d28d25 Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 104 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE Supported: replaces, from-change Server: SIPPER for PhonerLite Content-Length: 237 v=0 o=- 2923150685 3 IN IP4 192.168.210.111 s=SIPPER for PhonerLite c=IN IP4 192.168.210.111 t=0 0 m=audio 5062 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:2022430885 a=sendrecv <-------------> [Oct 18 10:04:03] DEBUG[23661] chan_sip.c: Call to peer '101' removed from call limit 2147483647 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK1080baed;rport=5060 [Oct 18 10:04:03] DEBUG[23661] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb663a450' [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as24b2d10e [Oct 18 10:04:03] DEBUG[23661] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb666b8d0' [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 3 [ 71]: To: ;tag=80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:04:03] VERBOSE[23661] chan_sip.c: Scheduling destruction of SIP dialog '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' in 32000 ms (Method: INVITE) [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 6 [ 39]: Contact: [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 8 [ 78]: Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 9 [ 32]: Supported: replaces, from-change [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 10 [ 29]: Server: SIPPER for PhonerLite [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 11 [ 19]: Content-Length: 237 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 12 [ 0]: [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 0 [ 3]: v=0 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 1 [ 39]: o=- 2923150685 3 IN IP4 192.168.210.111 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 2 [ 23]: s=SIPPER for PhonerLite [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 3 [ 24]: c=IN IP4 192.168.210.111 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 5 [ 26]: m=audio 5062 RTP/AVP 8 101 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 9 [ 17]: a=ssrc:2022430885 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 10 [ 10]: a=sendrecv [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: --- (12 headers 11 lines) --- [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: = Looking for Call ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 (Checking To) --From tag as24b2d10e --To-tag 80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:04:03] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 101 [Oct 18 10:04:03] DEBUG[23607] chan_sip.c: Checking device state for peer 101 [Oct 18 10:04:03] DEBUG[23607] devicestate.c: Changing state for SIP/101 - state 1 (Not in use) [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Acked pending invite 104 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #12 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Stopping retransmission on '205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060' of Request 104: Match Found [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: SIP response 200 to RE-invite on outgoing call 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23612] app_queue.c: Extension '101@LocalSets' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 18 10:04:03] DEBUG[23635] app_queue.c: Device 'SIP/101' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing session-level SDP o=- 2923150685 3 IN IP4 192.168.210.111... OK. [Oct 18 10:04:03] DEBUG[23667] ari/ari_websockets.c: Examining ARI event (length 184): { "type": "DeviceStateChanged", "application": "attendant", "timestamp": "2016-10-18T10:04:03.045+0200", "device_state": { "name": "SIP/101", "state": "NOT_INUSE" } } [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing session-level SDP s=SIPPER for PhonerLite... UNSUPPORTED OR FAILED. [Oct 18 10:04:03] DEBUG[23667] res_http_websocket.c: Writing websocket string of length 184 [Oct 18 10:04:03] DEBUG[23625] netsock2.c: Splitting '192.168.210.111' into... [Oct 18 10:04:03] DEBUG[23667] res_http_websocket.c: Writing websocket text frame, length 184 [Oct 18 10:04:03] DEBUG[23625] netsock2.c: ...host '192.168.210.111' and port ''. [Oct 18 10:04:03] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 101 [Oct 18 10:04:03] DEBUG[23607] chan_sip.c: Checking device state for peer 101 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.210.111... OK. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 18 10:04:03] DEBUG[23607] devicestate.c: Changing state for SIP/101 - state 1 (Not in use) [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Found RTP audio format 8 [Oct 18 10:04:03] DEBUG[23625] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb385047c [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Found RTP audio format 101 [Oct 18 10:04:03] DEBUG[23625] rtp_engine.c: Setting tx payload type 101 based on m type on 0xb385047c [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=ssrc:2022430885... UNSUPPORTED OR FAILED. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 18 10:04:03] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 526): { "type": "StasisEnd", "timestamp": "2016-10-18T10:04:03.046+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 526 [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 526 [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 18 10:04:03] DEBUG[23625] res_rtp_asterisk.c: Set role to CONTROLLING (0x8f033c8) [Oct 18 10:04:03] DEBUG[23625] res_rtp_asterisk.c: Set role failed; no ice instance (0x8f033c8) [Oct 18 10:04:03] DEBUG[23625] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8f033c8' [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Peer audio RTP is at port 192.168.210.111:5062 [Oct 18 10:04:03] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 8 (0xb6405d78) from 0xb385047c to 0x8f03574 [Oct 18 10:04:03] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 101 (0x8bedf80) from 0xb385047c to 0x8f03574 [Oct 18 10:04:03] DEBUG[23625] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x8f033c8' [Oct 18 10:04:03] DEBUG[23625] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8efbb68' [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Peer doesn't provide T.140 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: We're settling with these formats: (alaw) [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: We have an owner, now see if we need to change this call [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 18 10:04:03] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 582): { "value": "SUCCESS", "variable": "STASISSTATUS", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:03.043+0200", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Updating call counter for outgoing call [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 582 [Oct 18 10:04:03] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 290 [Oct 18 10:04:03] DEBUG[23607] chan_sip.c: Checking device state for peer 290 [Oct 18 10:04:03] DEBUG[23607] devicestate.c: Changing state for SIP/290 - state 2 (In use) [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Strict routing enforced for session 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 582 [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Transmitting (NAT) to 192.168.210.111:5060: ACK sip:290@192.168.210.111:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK2852725d;rport Max-Forwards: 70 From: "Anonymous" ;tag=as24b2d10e To: ;tag=80a63ee97693e611b2cfaca141d28d25 Contact: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 14.0.2 Content-Length: 0 --- [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Trying to put 'ACK sip:290' onto UDP socket destined for 192.168.210.111:5060 [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK2603fe63;rport=5060 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 104 INVITE User-Agent: Yealink SIP-T21P_E2 52.80.0.95 Content-Length: 0 <-------------> [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK2603fe63;rport=5060 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as3af1af94 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 3 [ 47]: To: ;tag=673443222 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 6 [ 42]: User-Agent: Yealink SIP-T21P_E2 52.80.0.95 [Oct 18 10:04:03] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 582): { "type": "ChannelDestroyed", "timestamp": "2016-10-18T10:04:03.044+0200", "cause": 16, "cause_txt": "Normal Clearing", "channel": { "id": "atom_asterisk-1476777778.0", "name": "SIP/101-00000000", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:02:58.554+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 582 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Oct 18 10:04:03] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 582 [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: --- (8 headers 0 lines) --- [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: = Looking for Call ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 (Checking To) --From tag as3af1af94 --To-tag 673443222 [Oct 18 10:04:03] DEBUG[23647] stasis/app.c: App 'attendant' not subscribed to channel 'atom_asterisk-1476777778.0' [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: *** SIP TIMER: Cancelling retransmission #13 - INVITE (got response) [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' Request 104: Found [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: SIP response 100 to RE-invite on outgoing call 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23676] res_rtp_asterisk.c: 0x8f03e60 -- Probation learning mode pass with source address 192.168.210.111:5062 [Oct 18 10:04:03] VERBOSE[23676] res_rtp_asterisk.c: 0x8f03e60 -- Probation passed - setting RTP source address to 192.168.210.111:5062 [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK2603fe63;rport=5060 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 104 INVITE Contact: Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE User-Agent: Yealink SIP-T21P_E2 52.80.0.95 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 215 v=0 o=- 20002 20004 IN IP4 192.168.210.40 s=SDP data c=IN IP4 192.168.210.40 t=0 0 m=audio 11788 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK2603fe63;rport=5060 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as3af1af94 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 3 [ 47]: To: ;tag=673443222 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 104 INVITE [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 6 [ 38]: Contact: [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 8 [115]: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 9 [ 42]: User-Agent: Yealink SIP-T21P_E2 52.80.0.95 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 10 [ 51]: Allow-Events: talk,hold,conference,refer,check-sync [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 11 [ 19]: Content-Length: 215 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 12 [ 0]: [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 0 [ 3]: v=0 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 1 [ 37]: o=- 20002 20004 IN IP4 192.168.210.40 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 2 [ 10]: s=SDP data [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.210.40 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 4 [ 5]: t=0 0 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 5 [ 27]: m=audio 11788 RTP/AVP 8 101 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 7 [ 10]: a=ptime:20 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Body 10 [ 10]: a=sendrecv [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: --- (12 headers 11 lines) --- [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: = Looking for Call ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 (Checking To) --From tag as3af1af94 --To-tag 673443222 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Acked pending invite 104 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Stopping retransmission on '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' of Request 104: Match Found [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: SIP response 200 to RE-invite on outgoing call 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing session-level SDP o=- 20002 20004 IN IP4 192.168.210.40... OK. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing session-level SDP s=SDP data... UNSUPPORTED OR FAILED. [Oct 18 10:04:03] DEBUG[23625] netsock2.c: Splitting '192.168.210.40' into... [Oct 18 10:04:03] DEBUG[23625] netsock2.c: ...host '192.168.210.40' and port ''. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.210.40... OK. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Found RTP audio format 8 [Oct 18 10:04:03] DEBUG[23625] rtp_engine.c: Setting tx payload type 8 based on m type on 0xb38504fc [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Found RTP audio format 101 [Oct 18 10:04:03] DEBUG[23625] rtp_engine.c: Setting tx payload type 101 based on m type on 0xb38504fc [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 18 10:04:03] DEBUG[23625] res_rtp_asterisk.c: Set role to CONTROLLING (0xb663a450) [Oct 18 10:04:03] DEBUG[23625] res_rtp_asterisk.c: Set role failed; no ice instance (0xb663a450) [Oct 18 10:04:03] DEBUG[23625] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb663a450' [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Peer audio RTP is at port 192.168.210.40:11788 [Oct 18 10:04:03] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 8 (0xb6401fd0) from 0xb38504fc to 0xb663a5fc [Oct 18 10:04:03] DEBUG[23625] rtp_engine.c: Copying tx payload mapping 101 (0x8bedf80) from 0xb38504fc to 0xb663a5fc [Oct 18 10:04:03] DEBUG[23625] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb663a450' [Oct 18 10:04:03] DEBUG[23625] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb666b8d0' [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Peer doesn't provide T.140 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: We're settling with these formats: (alaw) [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Updating call counter for outgoing call [Oct 18 10:04:03] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 101 [Oct 18 10:04:03] DEBUG[23607] chan_sip.c: Checking device state for peer 101 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Strict routing enforced for session 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23607] devicestate.c: Changing state for SIP/101 - state 1 (Not in use) [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Transmitting (NAT) to 192.168.210.40:5062: ACK sip:101@192.168.210.40:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK49abb699;rport Max-Forwards: 70 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Contact: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 14.0.2 Content-Length: 0 --- [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Trying to put 'ACK sip:101' onto UDP socket destined for 192.168.210.40:5062 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Strict routing enforced for session 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Reliably Transmitting (NAT) to 192.168.210.40:5062: BYE sip:101@192.168.210.40:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK2bec02f3;rport Max-Forwards: 70 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 105 BYE User-Agent: Asterisk PBX 14.0.2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #32 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Trying to put 'BYE sip:101' onto UDP socket destined for 192.168.210.40:5062 [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Scheduling destruction of SIP dialog '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' in 32000 ms (Method: INVITE) [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK2bec02f3;rport=5060 From: "Anonymous" ;tag=as3af1af94 To: ;tag=673443222 Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 CSeq: 105 BYE User-Agent: Yealink SIP-T21P_E2 52.80.0.95 Content-Length: 0 <-------------> [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK2bec02f3;rport=5060 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as3af1af94 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 3 [ 47]: To: ;tag=673443222 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 5 [ 13]: CSeq: 105 BYE [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 6 [ 42]: User-Agent: Yealink SIP-T21P_E2 52.80.0.95 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: --- (8 headers 0 lines) --- [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: = Looking for Call ID: 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 (Checking To) --From tag as3af1af94 --To-tag 673443222 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #32 [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Stopping retransmission on '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' of Request 105: Match Found [Oct 18 10:04:03] DEBUG[23625] chan_sip.c: Destroying SIP dialog 1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060 [Oct 18 10:04:03] VERBOSE[23625] chan_sip.c: Really destroying SIP dialog '1948c58e6fe7c992572f06284dd575a0@192.168.210.71:5060' Method: INVITE [Oct 18 10:04:03] DEBUG[23625] rtp_engine.c: Destroyed RTP instance '0xb663a450' [Oct 18 10:04:03] DEBUG[23625] rtp_engine.c: Destroyed RTP instance '0xb666b8d0' [Oct 18 10:04:06] DEBUG[23672] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 18 10:04:06] DEBUG[23596] threadpool.c: Destroying worker thread 9 [Oct 18 10:04:08] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:04:08] DEBUG[23625] netsock2.c: Splitting 'debian' into... [Oct 18 10:04:08] DEBUG[23625] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:04:08] DEBUG[23625] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:04:08] DEBUG[23625] acl.c: Attached to given IP address [Oct 18 10:04:09] DEBUG[23676] res_rtp_asterisk.c: Got RTCP report of 84 bytes [Oct 18 10:04:09] DEBUG[23676] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:04:09] DEBUG[23676] netsock2.c: Splitting 'debian' into... [Oct 18 10:04:09] DEBUG[23676] netsock2.c: ...host 'debian' and port ''. [Oct 18 10:04:09] DEBUG[23676] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [Oct 18 10:04:09] DEBUG[23676] acl.c: Attached to given IP address [Oct 18 10:04:10] DEBUG[23682] http.c: HTTP opening session. Top level [Oct 18 10:04:10] DEBUG[23682] http.c: HTTP Request URI is /ari/channels/atom_asterisk-1476777808.1 [Oct 18 10:04:10] DEBUG[23682] http.c: match request [ari/channels/atom_asterisk-1476777808.1] with handler [httpstatus] len 10 [Oct 18 10:04:10] DEBUG[23682] http.c: match request [ari/channels/atom_asterisk-1476777808.1] with handler [amanager] len 8 [Oct 18 10:04:10] DEBUG[23682] http.c: match request [ari/channels/atom_asterisk-1476777808.1] with handler [arawman] len 7 [Oct 18 10:04:10] DEBUG[23682] http.c: match request [ari/channels/atom_asterisk-1476777808.1] with handler [manager] len 7 [Oct 18 10:04:10] DEBUG[23682] http.c: match request [ari/channels/atom_asterisk-1476777808.1] with handler [rawman] len 6 [Oct 18 10:04:10] DEBUG[23682] http.c: match request [ari/channels/atom_asterisk-1476777808.1] with handler [static] len 6 [Oct 18 10:04:10] DEBUG[23682] http.c: match request [ari/channels/atom_asterisk-1476777808.1] with handler [amxml] len 5 [Oct 18 10:04:10] DEBUG[23682] http.c: match request [ari/channels/atom_asterisk-1476777808.1] with handler [mxml] len 4 [Oct 18 10:04:10] DEBUG[23682] http.c: match request [ari/channels/atom_asterisk-1476777808.1] with handler [ari] len 3 [Oct 18 10:04:10] DEBUG[23682] http.c: Match made with [ari] [Oct 18 10:04:10] DEBUG[23682] res_ari.c: Finding handler for channels [Oct 18 10:04:10] DEBUG[23682] res_ari.c: Checking events [Oct 18 10:04:10] DEBUG[23682] res_ari.c: Checking bridges [Oct 18 10:04:10] DEBUG[23682] res_ari.c: Checking channels [Oct 18 10:04:10] DEBUG[23682] res_ari.c: Got it! [Oct 18 10:04:10] DEBUG[23682] res_ari.c: Finding handler for atom_asterisk-1476777808.1 [Oct 18 10:04:10] DEBUG[23682] res_ari.c: Checking create [Oct 18 10:04:10] DEBUG[23682] res_ari.c: Checking channelId [Oct 18 10:04:10] DEBUG[23682] res_ari.c: Got it! [Oct 18 10:04:10] DEBUG[23682] channel.c: Soft-Hanging (0x20) up channel 'SIP/290-00000001' [Oct 18 10:04:10] DEBUG[23682] res_ari.c: Examining ARI response: 204 No Content [Oct 18 10:04:10] DEBUG[23682] http.c: HTTP keeping session open. status_code:204 [Oct 18 10:04:10] DEBUG[23676] bridge_channel.c: Setting 0xb663ca88(SIP/290-00000001) state from:0 to:1 [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 568): { "cause": 32, "soft": true, "type": "ChannelHangupRequest", "timestamp": "2016-10-18T10:04:10.253+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 568 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 568 [Oct 18 10:04:10] DEBUG[23676] bridge_channel.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: pulling 0xb663ca88(SIP/290-00000001) [Oct 18 10:04:10] VERBOSE[23676] bridge_channel.c: Channel SIP/290-00000001 left 'simple_bridge' stasis-bridge [Oct 18 10:04:10] DEBUG[23676] bridge_channel.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: 0xb663ca88(SIP/290-00000001) is leaving simple_bridge technology [Oct 18 10:04:10] DEBUG[23676] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 18 10:04:10] DEBUG[23676] bridge_native_rtp.c: Bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5' can not use native RTP bridge as two channels are required [Oct 18 10:04:10] DEBUG[23676] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 18 10:04:10] DEBUG[23676] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 18 10:04:10] DEBUG[23676] bridge.c: Chose bridge technology simple_bridge [Oct 18 10:04:10] DEBUG[23676] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5 is already using the new technology. [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 573): { "value": "", "variable": "BRIDGEPEER", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:10.253+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 573 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 573 [Oct 18 10:04:10] DEBUG[23676] stasis/control.c: atom_asterisk-1476777808.1, b1ecceb5-0432-40e6-b8c4-04dda2246cd5: Channel leaving bridge [Oct 18 10:04:10] DEBUG[23676] stasis/app.c: bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5': is 1 interested in attendant [Oct 18 10:04:10] DEBUG[23676] stasis/control.c: reason: Channel was departed from bridge [Oct 18 10:04:10] DEBUG[23671] stasis/control.c: atom_asterisk-1476777808.1: Channel departing bridge [Oct 18 10:04:10] DEBUG[23671] bridge.c: Waiting for 0xb663ca88(SIP/290-00000001) bridge thread to die. [Oct 18 10:04:10] DEBUG[23671] stasis/app.c: App 'attendant' not subscribed to bridge '__AST_BRIDGE_ALL_TOPIC' [Oct 18 10:04:10] DEBUG[23671] stasis/app.c: App 'attendant' not subscribed to channel 'atom_asterisk-1476777808.1' [Oct 18 10:04:10] DEBUG[23671] channel.c: Hanging up channel 'SIP/290-00000001' [Oct 18 10:04:10] DEBUG[23671] chan_sip.c: Hangup call SIP/290-00000001, SIP callid 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:04:10] DEBUG[23671] chan_sip.c: update_call_counter(290) - decrement call limit counter on hangup [Oct 18 10:04:10] DEBUG[23671] chan_sip.c: Updating call counter for outgoing call [Oct 18 10:04:10] DEBUG[23671] chan_sip.c: Call to peer '290' removed from call limit 2147483647 [Oct 18 10:04:10] DEBUG[23671] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8f033c8' [Oct 18 10:04:10] DEBUG[23671] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8efbb68' [Oct 18 10:04:10] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 290 [Oct 18 10:04:10] VERBOSE[23671] chan_sip.c: Scheduling destruction of SIP dialog '205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060' in 32000 ms (Method: INVITE) [Oct 18 10:04:10] DEBUG[23607] chan_sip.c: Checking device state for peer 290 [Oct 18 10:04:10] DEBUG[23607] devicestate.c: Changing state for SIP/290 - state 1 (Not in use) [Oct 18 10:04:10] DEBUG[23635] app_queue.c: Device 'SIP/290' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 18 10:04:10] DEBUG[23612] app_queue.c: Extension '290@LocalSets' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 18 10:04:10] DEBUG[23596] threadpool.c: Increasing threadpool stasis-core's size by 1 [Oct 18 10:04:10] DEBUG[23667] ari/ari_websockets.c: Examining ARI event (length 184): { "type": "DeviceStateChanged", "application": "attendant", "timestamp": "2016-10-18T10:04:10.257+0200", "device_state": { "name": "SIP/290", "state": "NOT_INUSE" } } [Oct 18 10:04:10] DEBUG[23667] res_http_websocket.c: Writing websocket string of length 184 [Oct 18 10:04:10] DEBUG[23667] res_http_websocket.c: Writing websocket text frame, length 184 [Oct 18 10:04:10] DEBUG[23671] chan_sip.c: Strict routing enforced for session 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:04:10] VERBOSE[23671] chan_sip.c: Reliably Transmitting (NAT) to 192.168.210.111:5060: BYE sip:290@192.168.210.111:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK749903dc;rport Max-Forwards: 70 From: "Anonymous" ;tag=as24b2d10e To: ;tag=80a63ee97693e611b2cfaca141d28d25 Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 105 BYE User-Agent: Asterisk PBX 14.0.2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Oct 18 10:04:10] DEBUG[23671] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #16 [Oct 18 10:04:10] DEBUG[23671] chan_sip.c: Trying to put 'BYE sip:290' onto UDP socket destined for 192.168.210.111:5060 [Oct 18 10:04:10] DEBUG[23682] http.c: HTTP idle timeout or peer closed connection. [Oct 18 10:04:10] DEBUG[23682] http.c: HTTP closing session. Top level [Oct 18 10:04:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.111:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK749903dc;rport=5060 From: "Anonymous" ;tag=as24b2d10e To: ;tag=80a63ee97693e611b2cfaca141d28d25 Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 CSeq: 105 BYE Contact: Server: SIPPER for PhonerLite Content-Length: 0 <-------------> [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK749903dc;rport=5060 [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Header 2 [ 66]: From: "Anonymous" ;tag=as24b2d10e [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Header 3 [ 71]: To: ;tag=80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Header 5 [ 13]: CSeq: 105 BYE [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Header 6 [ 39]: Contact: [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Header 7 [ 29]: Server: SIPPER for PhonerLite [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Oct 18 10:04:10] VERBOSE[23625] chan_sip.c: --- (9 headers 0 lines) --- [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: = Looking for Call ID: 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 (Checking To) --From tag as24b2d10e --To-tag 80a63ee97693e611b2cfaca141d28d25 [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16 [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Stopping retransmission on '205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060' of Request 105: Match Found [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Destroying SIP dialog 205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060 [Oct 18 10:04:10] VERBOSE[23625] chan_sip.c: Really destroying SIP dialog '205e83d36921e91544ba798f7cf2716f@192.168.210.71:5060' Method: INVITE [Oct 18 10:04:10] DEBUG[23625] rtp_engine.c: Destroyed RTP instance '0x8f033c8' [Oct 18 10:04:10] DEBUG[23625] rtp_engine.c: Destroyed RTP instance '0x8efbb68' [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 578): { "value": "", "variable": "BRIDGEPVTCALLID", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:10.253+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 578 [Oct 18 10:04:10] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 290 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 578 [Oct 18 10:04:10] DEBUG[23607] chan_sip.c: Checking device state for peer 290 [Oct 18 10:04:10] DEBUG[23607] devicestate.c: Changing state for SIP/290 - state 1 (Not in use) [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 757): { "type": "ChannelLeftBridge", "timestamp": "2016-10-18T10:04:10.254+0200", "bridge": { "id": "b1ecceb5-0432-40e6-b8c4-04dda2246cd5", "technology": "simple_bridge", "bridge_type": "mixing", "bridge_class": "stasis", "creator": "Stasis", "name": "", "channels": [] }, "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 757 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 757 [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 526): { "type": "StasisEnd", "timestamp": "2016-10-18T10:04:10.262+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 526 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 526 [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 582): { "value": "SUCCESS", "variable": "STASISSTATUS", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:10.256+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 582 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 582 [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 697): { "value": "ssrc=308686736;themssrc=2022430885;lp=851;rxjitter=0.000000;rxcount=1138;txjitter=0.000095;txcount=2;rlp=0;rtt=12326.868138", "variable": "RTPAUDIOQOS", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:10.256+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 697 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 697 [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 780): { "value": "minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;", "variable": "RTPAUDIOQOSJITTER", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:10.257+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 780 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 780 [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 762): { "value": "minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;", "variable": "RTPAUDIOQOSLOSS", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:10.257+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 762 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 762 [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 643): { "value": "minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;", "variable": "RTPAUDIOQOSRTT", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:10.257+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 643 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 643 [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 697): { "value": "ssrc=308686736;themssrc=2022430885;lp=851;rxjitter=0.000000;rxcount=1138;txjitter=0.000095;txcount=2;rlp=0;rtt=12326.868138", "variable": "RTPAUDIOQOS", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:10.257+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 697 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 697 [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 679): { "value": "ssrc=1278454548;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000", "variable": "RTPTEXTQOS", "type": "ChannelVarset", "timestamp": "2016-10-18T10:04:10.257+0200", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 679 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 679 [Oct 18 10:04:10] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 582): { "type": "ChannelDestroyed", "timestamp": "2016-10-18T10:04:10.258+0200", "cause": 16, "cause_txt": "Normal Clearing", "channel": { "id": "atom_asterisk-1476777808.1", "name": "SIP/290-00000001", "state": "Up", "caller": { "name": "", "number": "" }, "connected": { "name": "", "number": "" }, "accountcode": "", "dialplan": { "context": "LocalSets", "exten": "", "priority": 1 }, "creationtime": "2016-10-18T10:03:28.345+0200", "language": "en" }, "application": "attendant" } [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 582 [Oct 18 10:04:10] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 582 [Oct 18 10:04:10] DEBUG[23647] stasis/app.c: App 'attendant' not subscribed to channel 'atom_asterisk-1476777808.1' [Oct 18 10:04:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:04:10] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:04:14] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> <-------------> [Oct 18 10:04:16] DEBUG[23686] http.c: HTTP opening session. Top level [Oct 18 10:04:16] DEBUG[23686] http.c: HTTP Request URI is /ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5 [Oct 18 10:04:16] DEBUG[23686] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5] with handler [httpstatus] len 10 [Oct 18 10:04:16] DEBUG[23686] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5] with handler [amanager] len 8 [Oct 18 10:04:16] DEBUG[23686] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5] with handler [arawman] len 7 [Oct 18 10:04:16] DEBUG[23686] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5] with handler [manager] len 7 [Oct 18 10:04:16] DEBUG[23686] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5] with handler [rawman] len 6 [Oct 18 10:04:16] DEBUG[23686] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5] with handler [static] len 6 [Oct 18 10:04:16] DEBUG[23686] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5] with handler [amxml] len 5 [Oct 18 10:04:16] DEBUG[23686] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5] with handler [mxml] len 4 [Oct 18 10:04:16] DEBUG[23686] http.c: match request [ari/bridges/b1ecceb5-0432-40e6-b8c4-04dda2246cd5] with handler [ari] len 3 [Oct 18 10:04:16] DEBUG[23686] http.c: Match made with [ari] [Oct 18 10:04:16] DEBUG[23686] res_ari.c: Finding handler for bridges [Oct 18 10:04:16] DEBUG[23686] res_ari.c: Checking events [Oct 18 10:04:16] DEBUG[23686] res_ari.c: Checking bridges [Oct 18 10:04:16] DEBUG[23686] res_ari.c: Got it! [Oct 18 10:04:16] DEBUG[23686] res_ari.c: Finding handler for b1ecceb5-0432-40e6-b8c4-04dda2246cd5 [Oct 18 10:04:16] DEBUG[23686] res_ari.c: Checking bridgeId [Oct 18 10:04:16] DEBUG[23686] res_ari.c: Got it! [Oct 18 10:04:16] DEBUG[23686] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: telling all channels to leave the party [Oct 18 10:04:16] DEBUG[23686] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: dissolving bridge with cause 16(Normal Clearing) [Oct 18 10:04:16] DEBUG[23686] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: queueing action type:13 sub:1001 [Oct 18 10:04:16] DEBUG[23686] res_ari.c: Examining ARI response: 204 No Content [Oct 18 10:04:16] DEBUG[23606] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: actually destroying stasis bridge, nobody wants it anymore [Oct 18 10:04:16] DEBUG[23686] http.c: HTTP keeping session open. status_code:204 [Oct 18 10:04:16] DEBUG[23606] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling stasis bridge destructor [Oct 18 10:04:16] DEBUG[23606] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling simple_bridge technology stop [Oct 18 10:04:16] DEBUG[23606] bridge.c: Bridge b1ecceb5-0432-40e6-b8c4-04dda2246cd5: calling simple_bridge technology destructor [Oct 18 10:04:16] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 331): { "type": "BridgeDestroyed", "timestamp": "2016-10-18T10:04:16.845+0200", "bridge": { "id": "b1ecceb5-0432-40e6-b8c4-04dda2246cd5", "technology": "simple_bridge", "bridge_type": "mixing", "bridge_class": "stasis", "creator": "Stasis", "name": "", "channels": [] }, "application": "attendant" } [Oct 18 10:04:16] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 331 [Oct 18 10:04:16] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 331 [Oct 18 10:04:16] DEBUG[23647] stasis/app.c: bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5': is 0 interested in attendant [Oct 18 10:04:16] DEBUG[23647] stasis/app.c: bridge 'b1ecceb5-0432-40e6-b8c4-04dda2246cd5' unsubscribed from attendant [Oct 18 10:04:16] DEBUG[23686] http.c: HTTP idle timeout or peer closed connection. [Oct 18 10:04:16] DEBUG[23686] http.c: HTTP closing session. Top level [Oct 18 10:04:24] DEBUG[23689] http.c: HTTP opening session. Top level [Oct 18 10:04:24] DEBUG[23689] http.c: HTTP Request URI is /ari/bridges [Oct 18 10:04:24] DEBUG[23689] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 18 10:04:24] DEBUG[23689] http.c: match request [ari/bridges] with handler [amanager] len 8 [Oct 18 10:04:24] DEBUG[23689] http.c: match request [ari/bridges] with handler [arawman] len 7 [Oct 18 10:04:24] DEBUG[23689] http.c: match request [ari/bridges] with handler [manager] len 7 [Oct 18 10:04:24] DEBUG[23689] http.c: match request [ari/bridges] with handler [rawman] len 6 [Oct 18 10:04:24] DEBUG[23689] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 18 10:04:24] DEBUG[23689] http.c: match request [ari/bridges] with handler [amxml] len 5 [Oct 18 10:04:24] DEBUG[23689] http.c: match request [ari/bridges] with handler [mxml] len 4 [Oct 18 10:04:24] DEBUG[23689] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 18 10:04:24] DEBUG[23689] http.c: Match made with [ari] [Oct 18 10:04:24] DEBUG[23689] res_ari.c: Finding handler for bridges [Oct 18 10:04:24] DEBUG[23689] res_ari.c: Checking events [Oct 18 10:04:24] DEBUG[23689] res_ari.c: Checking bridges [Oct 18 10:04:24] DEBUG[23689] res_ari.c: Got it! [Oct 18 10:04:24] DEBUG[23689] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 18 10:04:24] DEBUG[23689] bridge_native_rtp.c: Bridge 'c4898e8c-bdee-4758-bbf4-f840bed5b59d' can not use native RTP bridge as two channels are required [Oct 18 10:04:24] DEBUG[23689] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 18 10:04:24] DEBUG[23689] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Oct 18 10:04:24] DEBUG[23689] bridge.c: Chose bridge technology simple_bridge [Oct 18 10:04:24] DEBUG[23689] bridge.c: Bridge c4898e8c-bdee-4758-bbf4-f840bed5b59d: calling simple_bridge technology constructor [Oct 18 10:04:24] DEBUG[23689] bridge.c: Bridge c4898e8c-bdee-4758-bbf4-f840bed5b59d: calling simple_bridge technology start [Oct 18 10:04:24] DEBUG[23689] res_ari.c: Examining ARI response: 200 OK Content-type: application/json { "id": "c4898e8c-bdee-4758-bbf4-f840bed5b59d", "technology": "simple_bridge", "bridge_type": "mixing", "bridge_class": "stasis", "creator": "Stasis", "name": "", "channels": [] } [Oct 18 10:04:24] DEBUG[23689] http.c: HTTP keeping session open. status_code:200 [Oct 18 10:04:24] DEBUG[23689] http.c: HTTP idle timeout or peer closed connection. [Oct 18 10:04:24] DEBUG[23689] http.c: HTTP closing session. Top level [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> REGISTER sip:192.168.210.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPj.rEDUw7WME2VIsCvbYSXoFZhzbl321LO Route: Max-Forwards: 70 From: ;tag=DBuTh-C1fCB7owh2sho9XA9SW.F4mafm To: Call-ID: ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx CSeq: 19312 REGISTER Authorization: Digest username="300", realm="atom_asterisk", nonce="661a319c", uri="sip:192.168.210.71", response="c96cf9ad2355d3e9670752e328ace159", algorithm=MD5 User-Agent: CSipSimple_ST-S450-19/r2459 Contact: Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 <-------------> [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 0 [ 35]: REGISTER sip:192.168.210.71 SIP/2.0 [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 1 [ 93]: Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPj.rEDUw7WME2VIsCvbYSXoFZhzbl321LO [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 2 [ 44]: Route: [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 3 [ 16]: Max-Forwards: 70 [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 4 [ 67]: From: ;tag=DBuTh-C1fCB7owh2sho9XA9SW.F4mafm [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 5 [ 28]: To: [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 6 [ 41]: Call-ID: ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 7 [ 20]: CSeq: 19312 REGISTER [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 8 [163]: Authorization: Digest username="300", realm="atom_asterisk", nonce="661a319c", uri="sip:192.168.210.71", response="c96cf9ad2355d3e9670752e328ace159", algorithm=MD5 [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 9 [ 39]: User-Agent: CSipSimple_ST-S450-19/r2459 [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 10 [ 43]: Contact: [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 11 [ 12]: Expires: 900 [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 12 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: --- (14 headers 0 lines) --- [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: = Looking for Call ID: ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx (Checking From) --From tag DBuTh-C1fCB7owh2sho9XA9SW.F4mafm --To-tag [Oct 18 10:04:27] DEBUG[23625] acl.c: For destination '192.168.210.110', our source address is '192.168.210.71'. [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.210.71:5060 [Oct 18 10:04:27] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:04:27] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (no NAT) [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Allocating new SIP dialog for ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx - REGISTER (No RTP) [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Initializing initreq for method REGISTER - callid ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx [Oct 18 10:04:27] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:04:27] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (no NAT) [Oct 18 10:04:27] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:04:27] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: <--- Transmitting (NAT) to 192.168.210.110:53027 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.210.110:53027;branch=z9hG4bKPj.rEDUw7WME2VIsCvbYSXoFZhzbl321LO;received=192.168.210.110;rport=53027 From: ;tag=DBuTh-C1fCB7owh2sho9XA9SW.F4mafm To: ;tag=as42da4fe7 Call-ID: ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx CSeq: 19312 REGISTER Server: Asterisk PBX 14.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="atom_asterisk", nonce="46032422" Content-Length: 0 <------------> [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.210.110:53027 [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: Scheduling destruction of SIP dialog 'ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx' in 32000 ms (Method: REGISTER) [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> REGISTER sip:192.168.210.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPj2RPheKA-p3tLzOyKNRU61hnuCugMWQOa Route: Max-Forwards: 70 From: ;tag=DBuTh-C1fCB7owh2sho9XA9SW.F4mafm To: Call-ID: ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx CSeq: 19313 REGISTER User-Agent: CSipSimple_ST-S450-19/r2459 Contact: Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="300", realm="atom_asterisk", nonce="46032422", uri="sip:192.168.210.71", response="74cf062fdb598f2c99cbfe48ccdb0eb8", algorithm=MD5 Content-Length: 0 <-------------> [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 0 [ 35]: REGISTER sip:192.168.210.71 SIP/2.0 [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 1 [ 93]: Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPj2RPheKA-p3tLzOyKNRU61hnuCugMWQOa [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 2 [ 44]: Route: [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 3 [ 16]: Max-Forwards: 70 [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 4 [ 67]: From: ;tag=DBuTh-C1fCB7owh2sho9XA9SW.F4mafm [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 5 [ 28]: To: [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 6 [ 41]: Call-ID: ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 7 [ 20]: CSeq: 19313 REGISTER [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 8 [ 39]: User-Agent: CSipSimple_ST-S450-19/r2459 [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 9 [ 43]: Contact: [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 10 [ 12]: Expires: 900 [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 11 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 12 [163]: Authorization: Digest username="300", realm="atom_asterisk", nonce="46032422", uri="sip:192.168.210.71", response="74cf062fdb598f2c99cbfe48ccdb0eb8", algorithm=MD5 [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: --- (14 headers 0 lines) --- [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: = Looking for Call ID: ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx (Checking From) --From tag DBuTh-C1fCB7owh2sho9XA9SW.F4mafm --To-tag [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Initializing initreq for method REGISTER - callid ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx [Oct 18 10:04:27] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:04:27] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (no NAT) [Oct 18 10:04:27] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:04:27] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: build_path: do not use Path headers [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: Saved useragent "CSipSimple_ST-S450-19/r2459" for peer 300 [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: <--- Transmitting (NAT) to 192.168.210.110:53027 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.210.110:53027;branch=z9hG4bKPj2RPheKA-p3tLzOyKNRU61hnuCugMWQOa;received=192.168.210.110;rport=53027 From: ;tag=DBuTh-C1fCB7owh2sho9XA9SW.F4mafm To: ;tag=as42da4fe7 Call-ID: ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx CSeq: 19313 REGISTER Server: Asterisk PBX 14.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 900 Contact: ;expires=900 Date: Tue, 18 Oct 2016 08:04:27 GMT Content-Length: 0 <------------> [Oct 18 10:04:27] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.210.110:53027 [Oct 18 10:04:27] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 300 [Oct 18 10:04:27] VERBOSE[23625] chan_sip.c: Scheduling destruction of SIP dialog 'ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx' in 32000 ms (Method: REGISTER) [Oct 18 10:04:27] DEBUG[23607] chan_sip.c: Checking device state for peer 300 [Oct 18 10:04:27] DEBUG[23607] devicestate.c: Changing state for SIP/300 - state 1 (Not in use) [Oct 18 10:04:27] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 312): { "type": "PeerStatusChange", "timestamp": "2016-10-18T10:04:27.996+0200", "endpoint": { "technology": "SIP", "resource": "300", "state": "online", "channel_ids": [] }, "peer": { "peer_status": "Registered", "address": "192.168.210.110:53027" }, "application": "attendant" } [Oct 18 10:04:27] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 312 [Oct 18 10:04:27] DEBUG[23635] app_queue.c: Device 'SIP/300' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 18 10:04:27] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 312 [Oct 18 10:04:27] DEBUG[23667] ari/ari_websockets.c: Examining ARI event (length 184): { "type": "DeviceStateChanged", "application": "attendant", "timestamp": "2016-10-18T10:04:27.997+0200", "device_state": { "name": "SIP/300", "state": "NOT_INUSE" } } [Oct 18 10:04:27] DEBUG[23667] res_http_websocket.c: Writing websocket string of length 184 [Oct 18 10:04:27] DEBUG[23667] res_http_websocket.c: Writing websocket text frame, length 184 [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjnpdyhaLdVoiEE73LHMKnUrBkSm0T1ki2 Max-Forwards: 70 From: ;tag=LF6eIwXczWvU3p1bTGkYA6mx537dM7iP To: Contact: Call-ID: mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB CSeq: 23994 SUBSCRIBE Route: Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: CSipSimple_ST-S450-19/r2459 Content-Length: 0 <-------------> [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 0 [ 40]: SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 1 [ 93]: Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjnpdyhaLdVoiEE73LHMKnUrBkSm0T1ki2 [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 3 [ 67]: From: ;tag=LF6eIwXczWvU3p1bTGkYA6mx537dM7iP [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 4 [ 28]: To: [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 5 [ 43]: Contact: [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 6 [ 41]: Call-ID: mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 7 [ 21]: CSeq: 23994 SUBSCRIBE [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 8 [ 44]: Route: [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 9 [ 22]: Event: message-summary [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 10 [ 13]: Expires: 3600 [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 11 [ 46]: Supported: replaces, 100rel, timer, norefersub [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 12 [ 42]: Accept: application/simple-message-summary [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 13 [ 46]: Allow-Events: presence, message-summary, refer [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 14 [ 39]: User-Agent: CSipSimple_ST-S450-19/r2459 [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: --- (16 headers 0 lines) --- [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: = Looking for Call ID: mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB (Checking From) --From tag LF6eIwXczWvU3p1bTGkYA6mx537dM7iP --To-tag [Oct 18 10:04:28] DEBUG[23625] acl.c: For destination '192.168.210.110', our source address is '192.168.210.71'. [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.210.71:5060 [Oct 18 10:04:28] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:04:28] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (no NAT) [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Allocating new SIP dialog for mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB - SUBSCRIBE (No RTP) [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: Creating new subscription [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB [Oct 18 10:04:28] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:04:28] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (no NAT) [Oct 18 10:04:28] VERBOSE[23625] sip/route.c: sip_route_dump: route/path hop: [Oct 18 10:04:28] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:04:28] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: Found peer '300' for '300' from 192.168.210.110:53027 [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: <--- Transmitting (NAT) to 192.168.210.110:53027 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.210.110:53027;branch=z9hG4bKPjnpdyhaLdVoiEE73LHMKnUrBkSm0T1ki2;received=192.168.210.110;rport=53027 From: ;tag=LF6eIwXczWvU3p1bTGkYA6mx537dM7iP To: ;tag=as6278d582 Call-ID: mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB CSeq: 23994 SUBSCRIBE Server: Asterisk PBX 14.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="atom_asterisk", nonce="60b56574" Content-Length: 0 <------------> [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.210.110:53027 [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: Scheduling destruction of SIP dialog 'mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB' in 32000 ms (Method: SUBSCRIBE) [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjRV62Ov8EIkN1XswJWJqzV9Stxb6Cq9uC Max-Forwards: 70 From: ;tag=LF6eIwXczWvU3p1bTGkYA6mx537dM7iP To: Contact: Call-ID: mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB CSeq: 23995 SUBSCRIBE Route: Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: CSipSimple_ST-S450-19/r2459 Authorization: Digest username="300", realm="atom_asterisk", nonce="60b56574", uri="sip:300@192.168.210.71", response="b90095ca28c67c55346df8087feace09", algorithm=MD5 Content-Length: 0 <-------------> [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 0 [ 40]: SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 1 [ 93]: Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjRV62Ov8EIkN1XswJWJqzV9Stxb6Cq9uC [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 3 [ 67]: From: ;tag=LF6eIwXczWvU3p1bTGkYA6mx537dM7iP [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 4 [ 28]: To: [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 5 [ 43]: Contact: [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 6 [ 41]: Call-ID: mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 7 [ 21]: CSeq: 23995 SUBSCRIBE [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 8 [ 44]: Route: [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 9 [ 22]: Event: message-summary [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 10 [ 13]: Expires: 3600 [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 11 [ 46]: Supported: replaces, 100rel, timer, norefersub [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 12 [ 42]: Accept: application/simple-message-summary [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 13 [ 46]: Allow-Events: presence, message-summary, refer [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 14 [ 39]: User-Agent: CSipSimple_ST-S450-19/r2459 [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 15 [167]: Authorization: Digest username="300", realm="atom_asterisk", nonce="60b56574", uri="sip:300@192.168.210.71", response="b90095ca28c67c55346df8087feace09", algorithm=MD5 [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Header 16 [ 17]: Content-Length: 0 [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: --- (17 headers 0 lines) --- [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: = Looking for Call ID: mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB (Checking From) --From tag LF6eIwXczWvU3p1bTGkYA6mx537dM7iP --To-tag [Oct 18 10:04:28] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:04:28] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:04:28] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:04:28] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Got a new subscription mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB (possibly with auth) or retransmission [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: Creating new subscription [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB [Oct 18 10:04:28] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:04:28] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (NAT) [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: build_route: Retaining previous route: [Oct 18 10:04:28] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:04:28] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: Found peer '300' for '300' from 192.168.210.110:53027 [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: <--- Transmitting (NAT) to 192.168.210.110:53027 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 192.168.210.110:53027;branch=z9hG4bKPjRV62Ov8EIkN1XswJWJqzV9Stxb6Cq9uC;received=192.168.210.110;rport=53027 From: ;tag=LF6eIwXczWvU3p1bTGkYA6mx537dM7iP To: ;tag=as6278d582 Call-ID: mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB CSeq: 23995 SUBSCRIBE Server: Asterisk PBX 14.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 192.168.210.110:53027 [Oct 18 10:04:28] NOTICE[23625] chan_sip.c: Received SIP subscribe for peer without mailbox: 300 [Oct 18 10:04:28] DEBUG[23625] chan_sip.c: Destroying SIP dialog mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB [Oct 18 10:04:28] VERBOSE[23625] chan_sip.c: Really destroying SIP dialog 'mGGMyQdfJJ-1n9hK12O3JQq6g2GkgtSB' Method: SUBSCRIBE [Oct 18 10:04:30] DEBUG[23683] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 18 10:04:30] DEBUG[23596] threadpool.c: Destroying worker thread 10 [Oct 18 10:04:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:04:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:04:47] DEBUG[23667] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 18 10:04:47] DEBUG[23596] threadpool.c: Destroying worker thread 8 [Oct 18 10:04:59] DEBUG[23625] chan_sip.c: Auto destroying SIP dialog 'ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx' [Oct 18 10:04:59] DEBUG[23625] chan_sip.c: Destroying SIP dialog ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx [Oct 18 10:04:59] VERBOSE[23625] chan_sip.c: Really destroying SIP dialog 'ZEOxUopKIQp3KcO8P03tHodSu1zuKWSx' Method: REGISTER [Oct 18 10:05:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:05:10] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:05:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:05:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:05:48] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> <-------------> [Oct 18 10:06:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:06:10] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:06:39] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.111:5060 ---> REGISTER sip:192.168.210.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK8086af5b7793e611b2cfaca141d28d25;rport From: "PhonerLite" ;tag=435475923 To: "PhonerLite" Call-ID: 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 CSeq: 9 REGISTER Contact: ;+sip.instance="" Authorization: Digest username="290", realm="atom_asterisk", nonce="4cac9833", uri="sip:192.168.210.71", response="a65527576773a6a132a3afbcca41d209", algorithm=MD5 Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE Max-Forwards: 70 Allow-Events: org.3gpp.nwinitdereg User-Agent: SIPPER for PhonerLite Supported: replaces, timer, from-change, gruu Expires: 900 Content-Length: 0 <-------------> [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 0 [ 35]: REGISTER sip:192.168.210.71 SIP/2.0 [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK8086af5b7793e611b2cfaca141d28d25;rport [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 2 [ 57]: From: "PhonerLite" ;tag=435475923 [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 3 [ 41]: To: "PhonerLite" [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 5 [ 16]: CSeq: 9 REGISTER [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 6 [103]: Contact: ;+sip.instance="" [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 7 [163]: Authorization: Digest username="290", realm="atom_asterisk", nonce="4cac9833", uri="sip:192.168.210.71", response="a65527576773a6a132a3afbcca41d209", algorithm=MD5 [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 8 [ 78]: Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 10 [ 34]: Allow-Events: org.3gpp.nwinitdereg [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 11 [ 33]: User-Agent: SIPPER for PhonerLite [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 12 [ 45]: Supported: replaces, timer, from-change, gruu [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 13 [ 12]: Expires: 900 [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 18 10:06:39] VERBOSE[23625] chan_sip.c: --- (15 headers 0 lines) --- [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: = Looking for Call ID: 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 (Checking From) --From tag 435475923 --To-tag [Oct 18 10:06:39] DEBUG[23625] acl.c: For destination '192.168.210.111', our source address is '192.168.210.71'. [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.210.71:5060 [Oct 18 10:06:39] DEBUG[23625] netsock2.c: Splitting '192.168.210.111:5060' into... [Oct 18 10:06:39] DEBUG[23625] netsock2.c: ...host '192.168.210.111' and port '5060'. [Oct 18 10:06:39] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.111:5060 (no NAT) [Oct 18 10:06:39] DEBUG[23625] chan_sip.c: Allocating new SIP dialog for 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 - REGISTER (No RTP) [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Initializing initreq for method REGISTER - callid 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 [Oct 18 10:06:40] DEBUG[23625] netsock2.c: Splitting '192.168.210.111:5060' into... [Oct 18 10:06:40] DEBUG[23625] netsock2.c: ...host '192.168.210.111' and port '5060'. [Oct 18 10:06:40] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.111:5060 (no NAT) [Oct 18 10:06:40] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:06:40] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:06:40] VERBOSE[23625] chan_sip.c: <--- Transmitting (NAT) to 192.168.210.111:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK8086af5b7793e611b2cfaca141d28d25;received=192.168.210.111;rport=5060 From: "PhonerLite" ;tag=435475923 To: "PhonerLite" ;tag=as68486d67 Call-ID: 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 CSeq: 9 REGISTER Server: Asterisk PBX 14.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="atom_asterisk", nonce="36321d92" Content-Length: 0 <------------> [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.210.111:5060 [Oct 18 10:06:40] VERBOSE[23625] chan_sip.c: Scheduling destruction of SIP dialog '8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111' in 32000 ms (Method: REGISTER) [Oct 18 10:06:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.111:5060 ---> REGISTER sip:192.168.210.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK8086af5b7793e611b2d0aca141d28d25;rport From: "PhonerLite" ;tag=435475923 To: "PhonerLite" Call-ID: 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 CSeq: 10 REGISTER Contact: ;+sip.instance="" Authorization: Digest username="290", realm="atom_asterisk", nonce="36321d92", uri="sip:192.168.210.71", response="7e5f17204aab5dcf54971269bd3b3534", algorithm=MD5 Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE Max-Forwards: 70 Allow-Events: org.3gpp.nwinitdereg User-Agent: SIPPER for PhonerLite Supported: replaces, timer, from-change, gruu Expires: 900 Content-Length: 0 <-------------> [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 0 [ 35]: REGISTER sip:192.168.210.71 SIP/2.0 [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK8086af5b7793e611b2d0aca141d28d25;rport [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 2 [ 57]: From: "PhonerLite" ;tag=435475923 [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 3 [ 41]: To: "PhonerLite" [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 4 [ 61]: Call-ID: 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 5 [ 17]: CSeq: 10 REGISTER [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 6 [103]: Contact: ;+sip.instance="" [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 7 [163]: Authorization: Digest username="290", realm="atom_asterisk", nonce="36321d92", uri="sip:192.168.210.71", response="7e5f17204aab5dcf54971269bd3b3534", algorithm=MD5 [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 8 [ 78]: Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 10 [ 34]: Allow-Events: org.3gpp.nwinitdereg [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 11 [ 33]: User-Agent: SIPPER for PhonerLite [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 12 [ 45]: Supported: replaces, timer, from-change, gruu [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 13 [ 12]: Expires: 900 [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Oct 18 10:06:40] VERBOSE[23625] chan_sip.c: --- (15 headers 0 lines) --- [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: = Looking for Call ID: 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 (Checking From) --From tag 435475923 --To-tag [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Initializing initreq for method REGISTER - callid 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 [Oct 18 10:06:40] DEBUG[23625] netsock2.c: Splitting '192.168.210.111:5060' into... [Oct 18 10:06:40] DEBUG[23625] netsock2.c: ...host '192.168.210.111' and port '5060'. [Oct 18 10:06:40] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.111:5060 (no NAT) [Oct 18 10:06:40] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:06:40] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Store REGISTER's src-IP:port for call routing. [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: build_path: do not use Path headers [Oct 18 10:06:40] VERBOSE[23625] chan_sip.c: Saved useragent "SIPPER for PhonerLite" for peer 290 [Oct 18 10:06:40] DEBUG[23596] threadpool.c: Increasing threadpool stasis-core's size by 1 [Oct 18 10:06:40] VERBOSE[23625] chan_sip.c: <--- Transmitting (NAT) to 192.168.210.111:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK8086af5b7793e611b2d0aca141d28d25;received=192.168.210.111;rport=5060 From: "PhonerLite" ;tag=435475923 To: "PhonerLite" ;tag=as68486d67 Call-ID: 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 CSeq: 10 REGISTER Server: Asterisk PBX 14.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 900 Contact: ;expires=900 Date: Tue, 18 Oct 2016 08:06:40 GMT Content-Length: 0 <------------> [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.210.111:5060 [Oct 18 10:06:40] VERBOSE[23625] chan_sip.c: Scheduling destruction of SIP dialog '8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111' in 32000 ms (Method: REGISTER) [Oct 18 10:06:40] DEBUG[23607] devicestate.c: No provider found, checking channel drivers for SIP - 290 [Oct 18 10:06:40] DEBUG[23607] chan_sip.c: Checking device state for peer 290 [Oct 18 10:06:40] DEBUG[23607] devicestate.c: Changing state for SIP/290 - state 1 (Not in use) [Oct 18 10:06:40] DEBUG[23647] ari/ari_websockets.c: Examining ARI event (length 311): { "type": "PeerStatusChange", "timestamp": "2016-10-18T10:06:40.003+0200", "endpoint": { "technology": "SIP", "resource": "290", "state": "online", "channel_ids": [] }, "peer": { "peer_status": "Registered", "address": "192.168.210.111:5060" }, "application": "attendant" } [Oct 18 10:06:40] DEBUG[23647] res_http_websocket.c: Writing websocket string of length 311 [Oct 18 10:06:40] DEBUG[23647] res_http_websocket.c: Writing websocket text frame, length 311 [Oct 18 10:06:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:06:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:07:00] DEBUG[23732] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 18 10:07:00] DEBUG[23596] threadpool.c: Destroying worker thread 11 [Oct 18 10:07:08] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> <-------------> [Oct 18 10:07:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:07:10] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjTxScAtIl.IlX7UxLtgEG-9bZtC3okfOm Max-Forwards: 70 From: ;tag=w-i8AqNgxGq.2RfR-s8D5832zbJm--XB To: Contact: Call-ID: keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y CSeq: 9993 SUBSCRIBE Route: Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: CSipSimple_ST-S450-19/r2459 Content-Length: 0 <-------------> [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 0 [ 40]: SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 1 [ 93]: Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjTxScAtIl.IlX7UxLtgEG-9bZtC3okfOm [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 3 [ 67]: From: ;tag=w-i8AqNgxGq.2RfR-s8D5832zbJm--XB [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 4 [ 28]: To: [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 5 [ 43]: Contact: [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 6 [ 41]: Call-ID: keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 7 [ 20]: CSeq: 9993 SUBSCRIBE [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 8 [ 44]: Route: [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 9 [ 22]: Event: message-summary [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 10 [ 13]: Expires: 3600 [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 11 [ 46]: Supported: replaces, 100rel, timer, norefersub [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 12 [ 42]: Accept: application/simple-message-summary [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 13 [ 46]: Allow-Events: presence, message-summary, refer [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 14 [ 39]: User-Agent: CSipSimple_ST-S450-19/r2459 [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: --- (16 headers 0 lines) --- [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: = Looking for Call ID: keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y (Checking From) --From tag w-i8AqNgxGq.2RfR-s8D5832zbJm--XB --To-tag [Oct 18 10:07:11] DEBUG[23625] acl.c: For destination '192.168.210.110', our source address is '192.168.210.71'. [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.210.71:5060 [Oct 18 10:07:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:07:11] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (no NAT) [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Allocating new SIP dialog for keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y - SUBSCRIBE (No RTP) [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: Creating new subscription [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y [Oct 18 10:07:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:07:11] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (no NAT) [Oct 18 10:07:11] VERBOSE[23625] sip/route.c: sip_route_dump: route/path hop: [Oct 18 10:07:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:07:11] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: Found peer '300' for '300' from 192.168.210.110:53027 [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: <--- Transmitting (NAT) to 192.168.210.110:53027 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.210.110:53027;branch=z9hG4bKPjTxScAtIl.IlX7UxLtgEG-9bZtC3okfOm;received=192.168.210.110;rport=53027 From: ;tag=w-i8AqNgxGq.2RfR-s8D5832zbJm--XB To: ;tag=as344fadff Call-ID: keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y CSeq: 9993 SUBSCRIBE Server: Asterisk PBX 14.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="atom_asterisk", nonce="32b5678f" Content-Length: 0 <------------> [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.210.110:53027 [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: Scheduling destruction of SIP dialog 'keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y' in 32000 ms (Method: SUBSCRIBE) [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjNFOWQuL14IEIpo8SKkWeuKFw713j0PaD Max-Forwards: 70 From: ;tag=w-i8AqNgxGq.2RfR-s8D5832zbJm--XB To: Contact: Call-ID: keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y CSeq: 9994 SUBSCRIBE Route: Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: CSipSimple_ST-S450-19/r2459 Authorization: Digest username="300", realm="atom_asterisk", nonce="32b5678f", uri="sip:300@192.168.210.71", response="b716ac6dcb9796c36dfe763d754a12bc", algorithm=MD5 Content-Length: 0 <-------------> [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 0 [ 40]: SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 1 [ 93]: Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjNFOWQuL14IEIpo8SKkWeuKFw713j0PaD [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 3 [ 67]: From: ;tag=w-i8AqNgxGq.2RfR-s8D5832zbJm--XB [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 4 [ 28]: To: [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 5 [ 43]: Contact: [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 6 [ 41]: Call-ID: keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 7 [ 20]: CSeq: 9994 SUBSCRIBE [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 8 [ 44]: Route: [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 9 [ 22]: Event: message-summary [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 10 [ 13]: Expires: 3600 [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 11 [ 46]: Supported: replaces, 100rel, timer, norefersub [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 12 [ 42]: Accept: application/simple-message-summary [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 13 [ 46]: Allow-Events: presence, message-summary, refer [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 14 [ 39]: User-Agent: CSipSimple_ST-S450-19/r2459 [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 15 [167]: Authorization: Digest username="300", realm="atom_asterisk", nonce="32b5678f", uri="sip:300@192.168.210.71", response="b716ac6dcb9796c36dfe763d754a12bc", algorithm=MD5 [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Header 16 [ 17]: Content-Length: 0 [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: --- (17 headers 0 lines) --- [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: = Looking for Call ID: keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y (Checking From) --From tag w-i8AqNgxGq.2RfR-s8D5832zbJm--XB --To-tag [Oct 18 10:07:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:07:11] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:07:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:07:11] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Got a new subscription keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y (possibly with auth) or retransmission [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: Creating new subscription [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y [Oct 18 10:07:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:07:11] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (NAT) [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: build_route: Retaining previous route: [Oct 18 10:07:11] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:07:11] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: Found peer '300' for '300' from 192.168.210.110:53027 [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: <--- Transmitting (NAT) to 192.168.210.110:53027 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 192.168.210.110:53027;branch=z9hG4bKPjNFOWQuL14IEIpo8SKkWeuKFw713j0PaD;received=192.168.210.110;rport=53027 From: ;tag=w-i8AqNgxGq.2RfR-s8D5832zbJm--XB To: ;tag=as344fadff Call-ID: keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y CSeq: 9994 SUBSCRIBE Server: Asterisk PBX 14.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 192.168.210.110:53027 [Oct 18 10:07:11] NOTICE[23625] chan_sip.c: Received SIP subscribe for peer without mailbox: 300 [Oct 18 10:07:11] DEBUG[23625] chan_sip.c: Destroying SIP dialog keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y [Oct 18 10:07:11] VERBOSE[23625] chan_sip.c: Really destroying SIP dialog 'keTQ3PcwjDlHvJfstIpa26ViEA4Bq27Y' Method: SUBSCRIBE [Oct 18 10:07:12] DEBUG[23625] chan_sip.c: Auto destroying SIP dialog '8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111' [Oct 18 10:07:12] DEBUG[23625] chan_sip.c: Destroying SIP dialog 8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111 [Oct 18 10:07:12] VERBOSE[23625] chan_sip.c: Really destroying SIP dialog '8098904E-7393-E611-B2C8-ACA141D28D25@192.168.210.111' Method: REGISTER [Oct 18 10:07:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:07:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:08:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:08:10] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:08:29] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> <-------------> [Oct 18 10:08:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:08:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:09:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:09:10] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:09:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:09:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:09:49] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> <-------------> [Oct 18 10:10:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:10:10] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:10:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:10:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:11:09] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> <-------------> [Oct 18 10:11:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:11:10] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:11:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:11:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:12:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:12:10] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjogJtd8p9mn056hPRRP5SSYDRemy5BJDs Max-Forwards: 70 From: ;tag=G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt To: Contact: Call-ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b CSeq: 29730 SUBSCRIBE Route: Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: CSipSimple_ST-S450-19/r2459 Content-Length: 0 <-------------> [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 0 [ 40]: SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 1 [ 93]: Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjogJtd8p9mn056hPRRP5SSYDRemy5BJDs [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 3 [ 67]: From: ;tag=G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 4 [ 28]: To: [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 5 [ 43]: Contact: [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 6 [ 41]: Call-ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 7 [ 21]: CSeq: 29730 SUBSCRIBE [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 8 [ 44]: Route: [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 9 [ 22]: Event: message-summary [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 10 [ 13]: Expires: 3600 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 11 [ 46]: Supported: replaces, 100rel, timer, norefersub [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 12 [ 42]: Accept: application/simple-message-summary [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 13 [ 46]: Allow-Events: presence, message-summary, refer [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 14 [ 39]: User-Agent: CSipSimple_ST-S450-19/r2459 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: --- (16 headers 0 lines) --- [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: = Looking for Call ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b (Checking From) --From tag G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt --To-tag [Oct 18 10:12:12] DEBUG[23625] acl.c: For destination '192.168.210.110', our source address is '192.168.210.71'. [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.210.71:5060 [Oct 18 10:12:12] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:12:12] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (no NAT) [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Allocating new SIP dialog for Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b - SUBSCRIBE (No RTP) [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Creating new subscription [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b [Oct 18 10:12:12] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:12:12] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (no NAT) [Oct 18 10:12:12] VERBOSE[23625] sip/route.c: sip_route_dump: route/path hop: [Oct 18 10:12:12] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:12:12] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Found peer '300' for '300' from 192.168.210.110:53027 [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: <--- Transmitting (NAT) to 192.168.210.110:53027 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.210.110:53027;branch=z9hG4bKPjogJtd8p9mn056hPRRP5SSYDRemy5BJDs;received=192.168.210.110;rport=53027 From: ;tag=G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt To: ;tag=as41354f7b Call-ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b CSeq: 29730 SUBSCRIBE Server: Asterisk PBX 14.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="atom_asterisk", nonce="257eddf3" Content-Length: 0 <------------> [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.210.110:53027 [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Scheduling destruction of SIP dialog 'Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b' in 32000 ms (Method: SUBSCRIBE) [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjogJtd8p9mn056hPRRP5SSYDRemy5BJDs Max-Forwards: 70 From: ;tag=G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt To: Contact: Call-ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b CSeq: 29730 SUBSCRIBE Route: Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: CSipSimple_ST-S450-19/r2459 Content-Length: 0 <-------------> [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 0 [ 40]: SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 1 [ 93]: Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPjogJtd8p9mn056hPRRP5SSYDRemy5BJDs [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 3 [ 67]: From: ;tag=G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 4 [ 28]: To: [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 5 [ 43]: Contact: [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 6 [ 41]: Call-ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 7 [ 21]: CSeq: 29730 SUBSCRIBE [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 8 [ 44]: Route: [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 9 [ 22]: Event: message-summary [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 10 [ 13]: Expires: 3600 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 11 [ 46]: Supported: replaces, 100rel, timer, norefersub [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 12 [ 42]: Accept: application/simple-message-summary [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 13 [ 46]: Allow-Events: presence, message-summary, refer [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 14 [ 39]: User-Agent: CSipSimple_ST-S450-19/r2459 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: --- (16 headers 0 lines) --- [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: = Looking for Call ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b (Checking From) --From tag G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt --To-tag [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Ignoring SIP message because of retransmit (SUBSCRIBE Seqno 29730, ours 29730) [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Got a new subscription Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b (possibly with auth) or retransmission [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Ignoring this SUBSCRIBE request [Oct 18 10:12:12] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:12:12] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Found peer '300' for '300' from 192.168.210.110:53027 [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: <--- Transmitting (NAT) to 192.168.210.110:53027 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.210.110:53027;branch=z9hG4bKPjogJtd8p9mn056hPRRP5SSYDRemy5BJDs;received=192.168.210.110;rport=53027 From: ;tag=G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt To: ;tag=as41354f7b Call-ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b CSeq: 29730 SUBSCRIBE Server: Asterisk PBX 14.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="atom_asterisk", nonce="257eddf3" Content-Length: 0 <------------> [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.210.110:53027 [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Scheduling destruction of SIP dialog 'Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b' in 32000 ms (Method: SUBSCRIBE) [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPj-ZfrqKljBA1q2MTELjbqE-JTQDNH7StO Max-Forwards: 70 From: ;tag=G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt To: Contact: Call-ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b CSeq: 29731 SUBSCRIBE Route: Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: CSipSimple_ST-S450-19/r2459 Authorization: Digest username="300", realm="atom_asterisk", nonce="257eddf3", uri="sip:300@192.168.210.71", response="ab879ff76c2c2bd3123bb5b396d3de76", algorithm=MD5 Content-Length: 0 <-------------> [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 0 [ 40]: SUBSCRIBE sip:300@192.168.210.71 SIP/2.0 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 1 [ 93]: Via: SIP/2.0/UDP 192.168.210.110:53027;rport;branch=z9hG4bKPj-ZfrqKljBA1q2MTELjbqE-JTQDNH7StO [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 3 [ 67]: From: ;tag=G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 4 [ 28]: To: [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 5 [ 43]: Contact: [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 6 [ 41]: Call-ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 7 [ 21]: CSeq: 29731 SUBSCRIBE [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 8 [ 44]: Route: [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 9 [ 22]: Event: message-summary [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 10 [ 13]: Expires: 3600 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 11 [ 46]: Supported: replaces, 100rel, timer, norefersub [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 12 [ 42]: Accept: application/simple-message-summary [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 13 [ 46]: Allow-Events: presence, message-summary, refer [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 14 [ 39]: User-Agent: CSipSimple_ST-S450-19/r2459 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 15 [167]: Authorization: Digest username="300", realm="atom_asterisk", nonce="257eddf3", uri="sip:300@192.168.210.71", response="ab879ff76c2c2bd3123bb5b396d3de76", algorithm=MD5 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Header 16 [ 17]: Content-Length: 0 [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: --- (17 headers 0 lines) --- [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: = Looking for Call ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b (Checking From) --From tag G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt --To-tag [Oct 18 10:12:12] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:12:12] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:12:12] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:12:12] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Got a new subscription Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b (possibly with auth) or retransmission [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Creating new subscription [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b [Oct 18 10:12:12] DEBUG[23625] netsock2.c: Splitting '192.168.210.110:53027' into... [Oct 18 10:12:12] DEBUG[23625] netsock2.c: ...host '192.168.210.110' and port '53027'. [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Sending to 192.168.210.110:53027 (NAT) [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: build_route: Retaining previous route: [Oct 18 10:12:12] DEBUG[23625] netsock2.c: Splitting '192.168.210.71' into... [Oct 18 10:12:12] DEBUG[23625] netsock2.c: ...host '192.168.210.71' and port ''. [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Found peer '300' for '300' from 192.168.210.110:53027 [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: <--- Transmitting (NAT) to 192.168.210.110:53027 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 192.168.210.110:53027;branch=z9hG4bKPj-ZfrqKljBA1q2MTELjbqE-JTQDNH7StO;received=192.168.210.110;rport=53027 From: ;tag=G7.8-G8SSrvOSiDKsvjXYy1GMsVMQubt To: ;tag=as41354f7b Call-ID: Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b CSeq: 29731 SUBSCRIBE Server: Asterisk PBX 14.0.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 192.168.210.110:53027 [Oct 18 10:12:12] NOTICE[23625] chan_sip.c: Received SIP subscribe for peer without mailbox: 300 [Oct 18 10:12:12] DEBUG[23625] chan_sip.c: Destroying SIP dialog Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b [Oct 18 10:12:12] VERBOSE[23625] chan_sip.c: Really destroying SIP dialog 'Y1fX9D-iziTBcyXS0JZHM0k6NrIV9Q7b' Method: SUBSCRIBE [Oct 18 10:12:29] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.110:53027 ---> <-------------> [Oct 18 10:12:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:12:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:13:10] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:13:10] DEBUG[23625] chan_sip.c: Header 0 [ 0]: [Oct 18 10:13:40] VERBOSE[23625] chan_sip.c: <--- SIP read from UDP:192.168.210.40:5062 ---> <-------------> [Oct 18 10:13:40] DEBUG[23625] chan_sip.c: Header 0 [ 0]: