[Oct 3 16:01:37] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> INVITE sip:222@10.10.10.103 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKcmDRoArlje7lJqbu Max-Forwards: 70 User-Agent: PA168S From: "alice" ;tag=pnZtIYPzVusub6w1 To: "222" Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 Contact: Authorization: Digest username="alice", realm="asterisk", nonce="3986e382", uri="sip:222@10.10.10.103", response="ed8d923c19c08202ca306e64ca7be07f", algorithm=MD5 CSeq: 2 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 188 v=0 o=- 39714279 91021611 IN IP4 10.10.10.2 s=SIP CALL c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Oct 3 16:01:37] VERBOSE[9957] chan_sip.c: --- (13 headers 9 lines) --- [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: Sending to 10.10.10.2:5060 (no NAT) [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: Using INVITE request as basis request - kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: Found peer 'alice' for 'alice' from 10.10.10.2:5060 [Oct 3 16:01:37] DEBUG[9957][C-00000001] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f3f3000b9b0' [Oct 3 16:01:37] DEBUG[9957][C-00000001] res_rtp_asterisk.c: Allocated port 16192 for RTP instance '0x7f3f3000b9b0' [Oct 3 16:01:37] DEBUG[9957][C-00000001] rtp_engine.c: RTP instance '0x7f3f3000b9b0' is setup and ready to go [Oct 3 16:01:37] DEBUG[9957][C-00000001] acl.c: Multiple addresses. Using the first only [Oct 3 16:01:37] DEBUG[9957][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f3f3000b9b0' [Oct 3 16:01:37] VERBOSE[9957][C-00000001] netsock2.c: Using SIP RTP CoS mark 5 [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Setting NAT on RTP to Off [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP o=- 39714279 91021611 IN IP4 10.10.10.2... OK. [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP s=SIP CALL... UNSUPPORTED OR FAILED. [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.10.10.2... OK. [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: Found RTP audio format 8 [Oct 3 16:01:37] DEBUG[9957][C-00000001] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f3ef02b5470 [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: Found RTP audio format 101 [Oct 3 16:01:37] DEBUG[9957][C-00000001] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f3ef02b5470 [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 3 16:01:37] DEBUG[9957][C-00000001] acl.c: For destination '10.10.10.2', our source address is '10.10.10.103'. [Oct 3 16:01:37] DEBUG[9957][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f3f3000b9b0' [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: Peer audio RTP is at port 10.10.10.2:1780 [Oct 3 16:01:37] DEBUG[9957][C-00000001] rtp_engine.c: Copying payload 8 (0x7f3f3000b0f8) from 0x7f3ef02b5470 to 0x7f3f3000bb78 [Oct 3 16:01:37] DEBUG[9957][C-00000001] rtp_engine.c: Copying payload 101 (0x7f3f30002a08) from 0x7f3ef02b5470 to 0x7f3f3000bb78 [Oct 3 16:01:37] DEBUG[9957][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f3f3000b9b0' [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: We're settling with these formats: (alaw) [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Checking SIP call limits for device alice [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Updating call counter for incoming call [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: Looking for 222 in local (domain 10.10.10.103) [Oct 3 16:01:37] DEBUG[9957][C-00000001] channel.c: Channel 0x7f3f30018f70 'SIP/alice-00000002' allocated [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: *** Our native formats are (alaw) [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: *** Joint capabilities are (alaw) [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: *** Our capabilities are (ulaw|alaw|gsm|h263) [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: This channel will not be able to handle video. [Oct 3 16:01:37] DEBUG[9916] threadpool.c: Increasing threadpool stasis-core's size by 1 [Oct 3 16:01:37] VERBOSE[9957][C-00000001] sip/route.c: sip_route_dump: route/path hop: [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: SIP/alice-00000002: New call is still down.... Trying... [Oct 3 16:01:37] VERBOSE[9957][C-00000001] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKcmDRoArlje7lJqbu;received=10.10.10.2 From: "alice" ;tag=pnZtIYPzVusub6w1 To: "222" Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:37] DEBUG[9927] chan_sip.c: Checking device state for peer alice [Oct 3 16:01:37] DEBUG[9927] devicestate.c: Changing state for SIP/alice - state 1 (Not in use) [Oct 3 16:01:37] DEBUG[10165][C-00000001] pbx.c: Launching 'Dial' [Oct 3 16:01:37] VERBOSE[10165][C-00000001] pbx.c: Executing [222@local:1] Dial("SIP/alice-00000002", "SIP/bob") in new stack [Oct 3 16:01:37] DEBUG[10165][C-00000001] pbx_lua.c: Looking up 222@local:-1 [Oct 3 16:01:37] DEBUG[10165][C-00000001] pbx_lua.c: Looking up 222@default:-1 [Oct 3 16:01:37] DEBUG[10165][C-00000001] pbx_lua.c: Looking up 222@demo:-1 [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: Asked to create a SIP channel with formats: (alaw) [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: Allocating new SIP dialog for 504d0e1a687435396cb6331c43c963c7@127.0.1.1:5060 - INVITE (No RTP) [Oct 3 16:01:37] DEBUG[10165][C-00000001] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f3f20007610' [Oct 3 16:01:37] DEBUG[10165][C-00000001] res_rtp_asterisk.c: Allocated port 16778 for RTP instance '0x7f3f20007610' [Oct 3 16:01:37] DEBUG[10165][C-00000001] rtp_engine.c: RTP instance '0x7f3f20007610' is setup and ready to go [Oct 3 16:01:37] DEBUG[10165][C-00000001] acl.c: Multiple addresses. Using the first only [Oct 3 16:01:37] DEBUG[10165][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f3f20007610' [Oct 3 16:01:37] VERBOSE[10165][C-00000001] netsock2.c: Using SIP RTP CoS mark 5 [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: Setting NAT on RTP to Off [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Oct 3 16:01:37] DEBUG[10165][C-00000001] acl.c: For destination '10.10.10.100', our source address is '10.10.10.103'. [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.10.10.103:5060 [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: Setting NAT on RTP to Off [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: SIP call-id changed from '504d0e1a687435396cb6331c43c963c7@127.0.1.1:5060' to '6047941670ac6f605ace08a861ed2402@10.10.10.103:5060' [Oct 3 16:01:37] DEBUG[10165][C-00000001] channel.c: Channel 0x7f3f20019630 'SIP/bob-00000003' allocated [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: *** Our native formats are (alaw) [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: *** Joint capabilities are (alaw) [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: *** Our capabilities are (ulaw|alaw|gsm|h263) [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: This channel will not be able to handle video. [Oct 3 16:01:37] DEBUG[10165][C-00000001] pbx_lua.c: Looking up @local:1 [Oct 3 16:01:37] DEBUG[10165][C-00000001] pbx_lua.c: Looking up @default:1 [Oct 3 16:01:37] DEBUG[10165][C-00000001] pbx_lua.c: Looking up @demo:1 [Oct 3 16:01:37] DEBUG[10165][C-00000001] pbx_lua.c: Looking up @demo:1 [Oct 3 16:01:37] DEBUG[10165][C-00000001] channel_internal_api.c: Channel Call ID changing from [C-00000001] to [C-00000001] [Oct 3 16:01:37] DEBUG[10165][C-00000001] rtp_engine.c: Copying payload 8 (0x7f3f3000b0f8) from 0x7f3f3000bb78 to 0x7f3f200077d8 [Oct 3 16:01:37] DEBUG[10165][C-00000001] rtp_engine.c: Copying payload 101 (0x7f3f30002a08) from 0x7f3f3000bb78 to 0x7f3f200077d8 [Oct 3 16:01:37] DEBUG[10165][C-00000001] rtp_engine.c: Seeded SDP of 'SIP/bob-00000003' with that of 'SIP/alice-00000002' [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: Outgoing Call for bob [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: Updating call counter for outgoing call [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: This call needs video offers, but there's no video support enabled! [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: ** Our capability: (alaw|ulaw|gsm|h263) Video flag: False Text flag: False [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: ** Our prefcodec: (alaw) [Oct 3 16:01:37] VERBOSE[10165][C-00000001] chan_sip.c: Audio is at 16778 [Oct 3 16:01:37] VERBOSE[10165][C-00000001] chan_sip.c: Adding codec alaw to SDP [Oct 3 16:01:37] VERBOSE[10165][C-00000001] chan_sip.c: Adding codec ulaw to SDP [Oct 3 16:01:37] VERBOSE[10165][C-00000001] chan_sip.c: Adding codec gsm to SDP [Oct 3 16:01:37] VERBOSE[10165][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw|gsm|h263) [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 [Oct 3 16:01:37] VERBOSE[10165][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.100:5060: INVITE sip:bob@10.10.10.100;line=63d76c6fa5f4ca2 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK2584d3fb Max-Forwards: 70 From: "alice" ;tag=as340f262a To: Contact: Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Date: Tue, 03 Oct 2017 14:01:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 313 v=0 o=root 1842940510 1842940510 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.103 t=0 0 m=audio 16778 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.10.10.100:5060 [Oct 3 16:01:37] VERBOSE[10165][C-00000001] app_dial.c: Called SIP/bob [Oct 3 16:01:37] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK2584d3fb From: "alice" ;tag=as340f262a To: Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 102 INVITE User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 3 16:01:37] VERBOSE[9957] chan_sip.c: --- (8 headers 0 lines) --- [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6047941670ac6f605ace08a861ed2402@10.10.10.103:5060' Request 102: Found [Oct 3 16:01:37] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK2584d3fb From: "alice" ;tag=as340f262a To: ;tag=1225084157 Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 102 INVITE Contact: User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 3 16:01:37] VERBOSE[9957] chan_sip.c: --- (9 headers 0 lines) --- [Oct 3 16:01:37] DEBUG[9957][C-00000001] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6047941670ac6f605ace08a861ed2402@10.10.10.103:5060' Request 102: Found [Oct 3 16:01:37] VERBOSE[9957][C-00000001] sip/route.c: sip_route_dump: route/path hop: [Oct 3 16:01:37] DEBUG[9927] chan_sip.c: Checking device state for peer bob [Oct 3 16:01:37] DEBUG[9927] devicestate.c: Changing state for SIP/bob - state 1 (Not in use) [Oct 3 16:01:37] VERBOSE[10165][C-00000001] app_dial.c: SIP/bob-00000003 is ringing [Oct 3 16:01:37] DEBUG[10165][C-00000001] rtp_engine.c: Setting early bridge SDP of 'SIP/alice-00000002' with that of 'SIP/bob-00000003' [Oct 3 16:01:37] VERBOSE[10165][C-00000001] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.2:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKcmDRoArlje7lJqbu;received=10.10.10.2 From: "alice" ;tag=pnZtIYPzVusub6w1 To: "222" ;tag=as237823d9 Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 3 16:01:37] DEBUG[10165][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK2584d3fb From: "alice" ;tag=as340f262a To: ;tag=1225084157 Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 182 v=0 o=bob 1635 2310 IN IP4 10.10.10.100 s=Talk c=IN IP4 10.10.10.100 t=0 0 m=audio 7078 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: --- (10 headers 9 lines) --- [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Acked pending invite 102 [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on '6047941670ac6f605ace08a861ed2402@10.10.10.103:5060' of Request 102: Match Found [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP o=bob 1635 2310 IN IP4 10.10.10.100... OK. [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.10.10.100... OK. [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Found RTP audio format 8 [Oct 3 16:01:38] DEBUG[9957][C-00000001] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f3ef02b46f0 [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Found RTP audio format 101 [Oct 3 16:01:38] DEBUG[9957][C-00000001] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f3ef02b46f0 [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 3 16:01:38] DEBUG[9957][C-00000001] acl.c: For destination '10.10.10.100', our source address is '10.10.10.103'. [Oct 3 16:01:38] DEBUG[9957][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f3f20007610' [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Peer audio RTP is at port 10.10.10.100:7078 [Oct 3 16:01:38] DEBUG[9957][C-00000001] rtp_engine.c: Copying payload 8 (0x7f3f3000e698) from 0x7f3ef02b46f0 to 0x7f3f200077d8 [Oct 3 16:01:38] DEBUG[9957][C-00000001] rtp_engine.c: Copying payload 101 (0x7f3f300034c8) from 0x7f3ef02b46f0 to 0x7f3f200077d8 [Oct 3 16:01:38] DEBUG[9957][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f3f20007610' [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: We're settling with these formats: (alaw) [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Updating call counter for outgoing call [Oct 3 16:01:38] VERBOSE[9957][C-00000001] sip/route.c: sip_route_dump: route/path hop: [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Strict routing enforced for session 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.100:5060 [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.10.10.100:5060: ACK sip:vita@10.10.10.100 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK20412f76 Max-Forwards: 70 From: "alice" ;tag=as340f262a To: ;tag=1225084157 Contact: Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 102 ACK User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Content-Length: 0 --- [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'ACK sip:vit' onto UDP socket destined for 10.10.10.100:5060 [Oct 3 16:01:38] DEBUG[9927] chan_sip.c: Checking device state for peer bob [Oct 3 16:01:38] DEBUG[9927] devicestate.c: Changing state for SIP/bob - state 1 (Not in use) [Oct 3 16:01:38] VERBOSE[10165][C-00000001] app_dial.c: SIP/bob-00000003 answered SIP/alice-00000002 [Oct 3 16:01:38] DEBUG[10165][C-00000001] rtp_engine.c: Setting early bridge SDP of 'SIP/alice-00000002' with that of 'SIP/bob-00000003' [Oct 3 16:01:38] DEBUG[9927] chan_sip.c: Checking device state for peer alice [Oct 3 16:01:38] DEBUG[9927] devicestate.c: Changing state for SIP/alice - state 1 (Not in use) [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: SIP answering channel: SIP/alice-00000002 [Oct 3 16:01:38] DEBUG[10165][C-00000001] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: ** Our prefcodec: (nothing) [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: Audio is at 16192 [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: Adding codec alaw to SDP [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: Setting framing on incoming call: 0 [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.10.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKcmDRoArlje7lJqbu;received=10.10.10.2 From: "alice" ;tag=pnZtIYPzVusub6w1 To: "222" ;tag=as237823d9 Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 216698452 216698452 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.103 t=0 0 m=audio 16192 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 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9' can not use native RTP bridge as two channels are required [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Chose bridge technology simple_bridge [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: calling simple_bridge technology constructor [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: calling simple_bridge technology start [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge_channel.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: 0x7f3f2001e540(SIP/bob-00000003) is joining [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge_channel.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: pushing 0x7f3f2001e540(SIP/bob-00000003) [Oct 3 16:01:38] VERBOSE[10166][C-00000001] bridge_channel.c: Channel SIP/bob-00000003 joined 'simple_bridge' basic-bridge <6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9> [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9' can not use native RTP bridge as two channels are required [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge.c: Chose bridge technology simple_bridge [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9 is already using the new technology. [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: 0x7f3f2001e540(SIP/bob-00000003) is joining simple_bridge technology [Oct 3 16:01:38] DEBUG[10166][C-00000001] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_channel.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: 0x7f3f200072d0(SIP/alice-00000002) is joining [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_channel.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: pushing 0x7f3f200072d0(SIP/alice-00000002) [Oct 3 16:01:38] VERBOSE[10165][C-00000001] bridge_channel.c: Channel SIP/alice-00000002 joined 'simple_bridge' basic-bridge <6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9> [Oct 3 16:01:38] DEBUG[9928] cdr.c: Finalized CDR for SIP/bob-00000003 - start 1507039297.409405 answer 1507039298.819964 end 1507039298.823413 dispo ANSWERED [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Checking compatability for channels 'SIP/bob-00000003' and 'SIP/alice-00000002' [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Chose bridge technology native_rtp [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: calling native_rtp technology constructor [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: moving 0x7f3f2001e540(SIP/bob-00000003) to dummy bridge temporarily [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: 0x7f3f2001e540(SIP/bob-00000003) is leaving simple_bridge technology (dummy) [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: calling simple_bridge technology stop [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: 0x7f3f200072d0(SIP/alice-00000002) is joining native_rtp technology [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Channel 'SIP/alice-00000002' is joining bridge tech [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Attaching hook data 0x7f3f20019310 to 'SIP/alice-00000002' [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: 0x7f3f2001e540(SIP/bob-00000003) is joining native_rtp technology [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Channel 'SIP/bob-00000003' is joining bridge tech [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Attaching hook data 0x7f3f2001bb80 to 'SIP/bob-00000003' [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Tech starting 'SIP/alice-00000002' and 'SIP/bob-00000003' with target 'none' [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: Deferring reinvite on SIP 'kjERd3dZtA5kWmHI@10.10.10.2' - It's audio will be redirected to IP 10.10.10.100:7078 [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: Sending reinvite on SIP '6047941670ac6f605ace08a861ed2402@10.10.10.103:5060' - It's audio soon redirected to IP 10.10.10.2:1780 [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: Strict routing enforced for session 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.100:5060 [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: ** Our native-bridge filtered capablity: (alaw) [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: ** Our prefcodec: (alaw) [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: Audio is at 16778 [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: Adding codec alaw to SDP [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: Adding codec ulaw to SDP [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: Adding codec gsm to SDP [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: Initializing already initialized SIP dialog 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 (presumably reinvite) [Oct 3 16:01:38] VERBOSE[10165][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.100:5060: INVITE sip:vita@10.10.10.100 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK03e6c97f Max-Forwards: 70 From: "alice" ;tag=as340f262a To: ;tag=1225084157 Contact: Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported 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: 310 v=0 o=root 1842940510 1842940511 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Oct 3 16:01:38] DEBUG[10165][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.10.10.100:5060 [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: calling native_rtp technology start [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: calling simple_bridge technology destructor [Oct 3 16:01:38] DEBUG[10165][C-00000001] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Checking compatability for channels 'SIP/alice-00000002' and 'SIP/bob-00000003' [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Chose bridge technology native_rtp [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9 is already using the new technology. [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Checking compatability for channels 'SIP/alice-00000002' and 'SIP/bob-00000003' [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge.c: Chose bridge technology native_rtp [Oct 3 16:01:38] DEBUG[10166][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9 is already using the new technology. [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK03e6c97f From: "alice" ;tag=as340f262a To: ;tag=1225084157 Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 103 INVITE User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: --- (8 headers 0 lines) --- [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6047941670ac6f605ace08a861ed2402@10.10.10.103:5060' Request 103: Found [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK03e6c97f From: "alice" ;tag=as340f262a To: ;tag=1225084157 Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 103 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 182 v=0 o=bob 1635 2311 IN IP4 10.10.10.100 s=Talk c=IN IP4 10.10.10.100 t=0 0 m=audio 7078 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: --- (10 headers 9 lines) --- [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Acked pending invite 103 [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on '6047941670ac6f605ace08a861ed2402@10.10.10.103:5060' of Request 103: Match Found [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP o=bob 1635 2311 IN IP4 10.10.10.100... OK. [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.10.10.100... OK. [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Found RTP audio format 8 [Oct 3 16:01:38] DEBUG[9957][C-00000001] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f3ef02b46f0 [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Found RTP audio format 101 [Oct 3 16:01:38] DEBUG[9957][C-00000001] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f3ef02b46f0 [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 3 16:01:38] DEBUG[9957][C-00000001] acl.c: For destination '10.10.10.100', our source address is '10.10.10.103'. [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Peer audio RTP is at port 10.10.10.100:7078 [Oct 3 16:01:38] DEBUG[9957][C-00000001] rtp_engine.c: Copying payload 8 (0x7f3f30005618) from 0x7f3ef02b46f0 to 0x7f3f200077d8 [Oct 3 16:01:38] DEBUG[9957][C-00000001] rtp_engine.c: Copying payload 101 (0x7f3f30003508) from 0x7f3ef02b46f0 to 0x7f3f200077d8 [Oct 3 16:01:38] DEBUG[9957][C-00000001] acl.c: Attached to given IP address [Oct 3 16:01:38] DEBUG[9957][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f3f20007610' [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: We're settling with these formats: (alaw) [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Updating call counter for outgoing call [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Strict routing enforced for session 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.100:5060 [Oct 3 16:01:38] VERBOSE[9957][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.10.10.100:5060: ACK sip:vita@10.10.10.100 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK49c81151 Max-Forwards: 70 From: "alice" ;tag=as340f262a To: ;tag=1225084157 Contact: Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 103 ACK User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Content-Length: 0 --- [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'ACK sip:vit' onto UDP socket destined for 10.10.10.100:5060 [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Tech starting 'SIP/alice-00000002' and 'SIP/bob-00000003' with target 'SIP/alice-00000002' [Oct 3 16:01:38] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Sending 'SIP/alice-00000002' back to remote [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: Retransmitting #1 (no NAT) to 10.10.10.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKcmDRoArlje7lJqbu;received=10.10.10.2 From: "alice" ;tag=pnZtIYPzVusub6w1 To: "222" ;tag=as237823d9 Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 216698452 216698452 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.103 t=0 0 m=audio 16192 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 3 16:01:38] DEBUG[9957] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> ACK sip:222@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKdLO8Zq4zOiRFiKRU Max-Forwards: 70 User-Agent: PA168S From: "alice" ;tag=pnZtIYPzVusub6w1 To: "222" ;tag=as237823d9 Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 Contact: Authorization: Digest username="alice", realm="asterisk", nonce="3986e382", uri="sip:222@10.10.10.103:5060", response="e12827066dd1fcd44ba2f6cd649393e0", algorithm=MD5 CSeq: 2 ACK Content-Length: 0 <-------------> [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: --- (11 headers 0 lines) --- [Oct 3 16:01:38] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on 'kjERd3dZtA5kWmHI@10.10.10.2' of Response 2: Match Found [Oct 3 16:01:38] DEBUG[9957] chan_sip.c: Sending pending reinvite on 'kjERd3dZtA5kWmHI@10.10.10.2' [Oct 3 16:01:38] DEBUG[9957] chan_sip.c: Strict routing enforced for session kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 3 16:01:38] DEBUG[9957] chan_sip.c: ** Our native-bridge filtered capablity: (alaw) [Oct 3 16:01:38] DEBUG[9957] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 3 16:01:38] DEBUG[9957] chan_sip.c: ** Our prefcodec: (nothing) [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: Audio is at 16192 [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: Adding codec alaw to SDP [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 3 16:01:38] DEBUG[9957] chan_sip.c: -- Done with adding codecs to SDP [Oct 3 16:01:38] DEBUG[9957] chan_sip.c: Setting framing on incoming call: 0 [Oct 3 16:01:38] DEBUG[9957] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 3 16:01:38] DEBUG[9957] chan_sip.c: Initializing already initialized SIP dialog kjERd3dZtA5kWmHI@10.10.10.2 (presumably reinvite) [Oct 3 16:01:38] VERBOSE[9957] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.2:5060: INVITE sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK20d4335f Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 102 INVITE User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported 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: 263 v=0 o=root 216698452 216698453 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.100 t=0 0 m=audio 7078 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Oct 3 16:01:38] DEBUG[9957] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:39] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> ACK sip:222@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKdLO8Zq4zOiRFiKRU Max-Forwards: 70 User-Agent: PA168S From: "alice" ;tag=pnZtIYPzVusub6w1 To: "222" ;tag=as237823d9 Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 Contact: Authorization: Digest username="alice", realm="asterisk", nonce="3986e382", uri="sip:222@10.10.10.103:5060", response="e12827066dd1fcd44ba2f6cd649393e0", algorithm=MD5 CSeq: 2 ACK Content-Length: 0 <-------------> [Oct 3 16:01:39] VERBOSE[9957] chan_sip.c: --- (11 headers 0 lines) --- [Oct 3 16:01:39] VERBOSE[9957] chan_sip.c: Retransmitting #1 (no NAT) to 10.10.10.2:5060: INVITE sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK20d4335f Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 102 INVITE User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported 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: 263 v=0 o=root 216698452 216698453 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.100 t=0 0 m=audio 7078 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Oct 3 16:01:39] DEBUG[9957] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:39] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK20d4335f Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 102 INVITE From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Supported: replaces Content-Type: application/sdp Content-Length: 188 v=0 o=- 95856702 40645390 IN IP4 10.10.10.2 s=SIP CALL c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Oct 3 16:01:39] VERBOSE[9957] chan_sip.c: --- (11 headers 9 lines) --- [Oct 3 16:01:39] DEBUG[9957][C-00000001] chan_sip.c: Acked pending invite 102 [Oct 3 16:01:39] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on 'kjERd3dZtA5kWmHI@10.10.10.2' of Request 102: Match Found [Oct 3 16:01:39] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 3 16:01:39] DEBUG[9957][C-00000001] chan_sip.c: Call kjERd3dZtA5kWmHI@10.10.10.2 responded to our reinvite without changing SDP version; ignoring SDP. [Oct 3 16:01:39] DEBUG[9957][C-00000001] chan_sip.c: Updating call counter for incoming call [Oct 3 16:01:39] DEBUG[9957][C-00000001] chan_sip.c: Strict routing enforced for session kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:39] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:39] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 3 16:01:39] VERBOSE[9957][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.10.10.2:5060: ACK sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK5876d76e Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 102 ACK User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Content-Length: 0 --- [Oct 3 16:01:39] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'ACK sip:ali' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:39] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Tech starting 'SIP/alice-00000002' and 'SIP/bob-00000003' with target 'SIP/bob-00000003' [Oct 3 16:01:39] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Sending 'SIP/bob-00000003' back to remote [Oct 3 16:01:39] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK20d4335f Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 102 INVITE From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Supported: replaces Content-Type: application/sdp Content-Length: 188 v=0 o=- 95856702 40645390 IN IP4 10.10.10.2 s=SIP CALL c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Oct 3 16:01:39] VERBOSE[9957] chan_sip.c: --- (11 headers 9 lines) --- [Oct 3 16:01:39] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on 'kjERd3dZtA5kWmHI@10.10.10.2' of Request 102: Match Not Found [Oct 3 16:01:39] DEBUG[9957][C-00000001] chan_sip.c: Strict routing enforced for session kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:39] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:39] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 3 16:01:39] VERBOSE[9957][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.10.10.2:5060: ACK sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK0101ef40 Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 102 ACK User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Content-Length: 0 --- [Oct 3 16:01:39] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'ACK sip:ali' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:40] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.1:5060 ---> <-------------> [Oct 3 16:01:41] DEBUG[9957] chan_sip.c: Allocating new SIP dialog for 41f863e52202ed0872f84b4a23602c9c@127.0.1.1:5060 - OPTIONS (No RTP) [Oct 3 16:01:41] DEBUG[9957] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Oct 3 16:01:41] DEBUG[9957] acl.c: For destination '10.10.10.100', our source address is '10.10.10.103'. [Oct 3 16:01:41] DEBUG[9957] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.10.10.103:5060 [Oct 3 16:01:41] DEBUG[9957] chan_sip.c: SIP call-id changed from '41f863e52202ed0872f84b4a23602c9c@127.0.1.1:5060' to '28ac5ca87803d67a53ae8a78094ef6a1@10.10.10.103:5060' [Oct 3 16:01:41] DEBUG[9957] chan_sip.c: Initializing initreq for method OPTIONS - callid 28ac5ca87803d67a53ae8a78094ef6a1@10.10.10.103:5060 [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.100:5060: OPTIONS sip:bob@10.10.10.100;line=63d76c6fa5f4ca2 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK0afcf6d5 Max-Forwards: 70 From: "asterisk" ;tag=as056da066 To: Contact: Call-ID: 28ac5ca87803d67a53ae8a78094ef6a1@10.10.10.103:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Date: Tue, 03 Oct 2017 14:01:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Oct 3 16:01:41] DEBUG[9957] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.10.10.100:5060 [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK0afcf6d5 From: "asterisk" ;tag=as056da066 To: ;tag=1756871875 Call-ID: 28ac5ca87803d67a53ae8a78094ef6a1@10.10.10.103:5060 CSeq: 102 OPTIONS Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO Accept: application/sdp User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: --- (10 headers 0 lines) --- [Oct 3 16:01:41] DEBUG[9957] chan_sip.c: Stopping retransmission on '28ac5ca87803d67a53ae8a78094ef6a1@10.10.10.103:5060' of Request 102: Match Found [Oct 3 16:01:41] DEBUG[9957] chan_sip.c: Destroying SIP dialog 28ac5ca87803d67a53ae8a78094ef6a1@10.10.10.103:5060 [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: Really destroying SIP dialog '28ac5ca87803d67a53ae8a78094ef6a1@10.10.10.103:5060' Method: OPTIONS [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> INVITE sip:alice@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport;branch=z9hG4bK678291861 From: ;tag=1225084157 To: "alice" ;tag=as340f262a Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 2 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Subject: Call on hold Content-Length: 212 v=0 o=bob 1635 2312 IN IP4 10.10.10.100 s=Talk c=IN IP4 0.0.0.0 t=0 0 m=audio 7078 RTP/AVP 8 101 c=IN IP4 10.10.10.100 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendonly <-------------> [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: --- (13 headers 11 lines) --- [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Sending to 10.10.10.100:5060 (no NAT) [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP o=bob 1635 2312 IN IP4 10.10.10.100... OK. [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 0.0.0.0... OK. [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Found RTP audio format 8 [Oct 3 16:01:41] DEBUG[9957][C-00000001] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f3ef02b5470 [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Found RTP audio format 101 [Oct 3 16:01:41] DEBUG[9957][C-00000001] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f3ef02b5470 [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 10.10.10.100... OK. [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 3 16:01:41] DEBUG[9957][C-00000001] acl.c: For destination '10.10.10.100', our source address is '10.10.10.103'. [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Peer audio RTP is at port 10.10.10.100:7078 [Oct 3 16:01:41] DEBUG[9957][C-00000001] rtp_engine.c: Copying payload 8 (0x7f3f30018468) from 0x7f3ef02b5470 to 0x7f3f200077d8 [Oct 3 16:01:41] DEBUG[9957][C-00000001] rtp_engine.c: Copying payload 101 (0x7f3f30018d28) from 0x7f3ef02b5470 to 0x7f3f200077d8 [Oct 3 16:01:41] DEBUG[9957][C-00000001] acl.c: Attached to given IP address [Oct 3 16:01:41] DEBUG[9957][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f3f20007610' [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: We're settling with these formats: (alaw) [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Got a SIP re-invite for call 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: SIP/bob-00000003: This call is UP.... [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK678291861;received=10.10.10.100;rport=5060 From: ;tag=1225084157 To: "alice" ;tag=as340f262a Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.10.10.100:5060 [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: ** Our native-bridge filtered capablity: (alaw) [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: ** Our prefcodec: (alaw) [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Audio is at 16778 [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Adding codec alaw to SDP [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Adding codec ulaw to SDP [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Adding codec gsm to SDP [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Setting framing on incoming call: 0 [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.10.10.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK678291861;received=10.10.10.100;rport=5060 From: ;tag=1225084157 To: "alice" ;tag=as340f262a Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 2 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 310 v=0 o=root 1842940510 1842940512 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=recvonly <------------> [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.100:5060 [Oct 3 16:01:41] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Tech stopping 'SIP/alice-00000002' and 'SIP/bob-00000003' with target 'SIP/alice-00000002' [Oct 3 16:01:41] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Bringing back 'SIP/alice-00000002' to us [Oct 3 16:01:41] DEBUG[10165][C-00000001] chan_sip.c: Sending reinvite on SIP 'kjERd3dZtA5kWmHI@10.10.10.2' - It's audio soon redirected to IP 10.10.10.103:5060 [Oct 3 16:01:41] DEBUG[10165][C-00000001] acl.c: Attached to given IP address [Oct 3 16:01:41] DEBUG[10165][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f3f3000b9b0' [Oct 3 16:01:41] DEBUG[10165][C-00000001] chan_sip.c: Strict routing enforced for session kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:41] VERBOSE[10165][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:41] VERBOSE[10165][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 3 16:01:41] DEBUG[10165][C-00000001] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 3 16:01:41] DEBUG[10165][C-00000001] chan_sip.c: ** Our prefcodec: (nothing) [Oct 3 16:01:41] VERBOSE[10165][C-00000001] chan_sip.c: Audio is at 16192 [Oct 3 16:01:41] VERBOSE[10165][C-00000001] chan_sip.c: Adding codec alaw to SDP [Oct 3 16:01:41] VERBOSE[10165][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 3 16:01:41] DEBUG[10165][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Oct 3 16:01:41] DEBUG[10165][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 3 16:01:41] DEBUG[10165][C-00000001] chan_sip.c: Initializing already initialized SIP dialog kjERd3dZtA5kWmHI@10.10.10.2 (presumably reinvite) [Oct 3 16:01:41] VERBOSE[10165][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.2:5060: INVITE sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK2ecf5886 Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 103 INVITE User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported 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: 264 v=0 o=root 216698452 216698454 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.103 t=0 0 m=audio 16192 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 3 16:01:41] DEBUG[10165][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:41] DEBUG[10165][C-00000001] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/alice-00000002' and 'SIP/bob-00000003' - media will flow through Asterisk core [Oct 3 16:01:41] DEBUG[10165][C-00000001] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 3 16:01:41] VERBOSE[10165][C-00000001] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/alice-00000002' [Oct 3 16:01:41] DEBUG[10165][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 3 16:01:41] DEBUG[10165][C-00000001] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 3 16:01:41] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Tech starting 'SIP/alice-00000002' and 'SIP/bob-00000003' with target 'SIP/alice-00000002' [Oct 3 16:01:41] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Sending 'SIP/alice-00000002' back to remote [Oct 3 16:01:41] DEBUG[10165][C-00000001] channel.c: Channel SIP/alice-00000002 setting write format path: slin -> alaw [Oct 3 16:01:41] DEBUG[10165][C-00000001] res_musiconhold.c: SIP/alice-00000002 Opened file 0 '/var/lib/asterisk/moh/macroform-the_simplicity' [Oct 3 16:01:41] DEBUG[10165][C-00000001] res_rtp_asterisk.c: Ooh, format changed from none to alaw [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> ACK sip:alice@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport;branch=z9hG4bK2020465640 From: ;tag=1225084157 To: "alice" ;tag=as340f262a Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 2 ACK Contact: Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: --- (10 headers 0 lines) --- [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on '6047941670ac6f605ace08a861ed2402@10.10.10.103:5060' of Response 2: Match Found [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK2ecf5886 Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 103 INVITE From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Supported: replaces Content-Type: application/sdp Content-Length: 188 v=0 o=- 63909098 67924097 IN IP4 10.10.10.2 s=SIP CALL c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: --- (11 headers 9 lines) --- [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Acked pending invite 103 [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on 'kjERd3dZtA5kWmHI@10.10.10.2' of Request 103: Match Found [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Call kjERd3dZtA5kWmHI@10.10.10.2 responded to our reinvite without changing SDP version; ignoring SDP. [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Updating call counter for incoming call [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Strict routing enforced for session kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.10.10.2:5060: ACK sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK6114de1d Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 103 ACK User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Content-Length: 0 --- [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'ACK sip:ali' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:41] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Tech starting 'SIP/alice-00000002' and 'SIP/bob-00000003' with target 'SIP/bob-00000003' [Oct 3 16:01:41] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Sending 'SIP/bob-00000003' back to remote [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK2ecf5886 Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 103 INVITE From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Supported: replaces Content-Type: application/sdp Content-Length: 188 v=0 o=- 63909098 67924097 IN IP4 10.10.10.2 s=SIP CALL c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Oct 3 16:01:41] VERBOSE[9957] chan_sip.c: --- (11 headers 9 lines) --- [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on 'kjERd3dZtA5kWmHI@10.10.10.2' of Request 103: Match Not Found [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Strict routing enforced for session kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 3 16:01:41] VERBOSE[9957][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.10.10.2:5060: ACK sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK79fd4f2f Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 103 ACK User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Content-Length: 0 --- [Oct 3 16:01:41] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'ACK sip:ali' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:41] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:42] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:43] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:44] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:45] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] DEBUG[10166][C-00000001] res_rtp_asterisk.c: No remote address on RTP instance '0x7f3f20007610' so dropping frame [Oct 3 16:01:46] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> INVITE sip:alice@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport;branch=z9hG4bK1191253294 From: ;tag=1225084157 To: "alice" ;tag=as340f262a Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 3 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Subject: Call resuming Content-Length: 182 v=0 o=bob 1635 2313 IN IP4 10.10.10.100 s=Talk c=IN IP4 10.10.10.100 t=0 0 m=audio 7078 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Oct 3 16:01:46] VERBOSE[9957] chan_sip.c: --- (13 headers 9 lines) --- [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Sending to 10.10.10.100:5060 (no NAT) [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP o=bob 1635 2313 IN IP4 10.10.10.100... OK. [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.10.10.100... OK. [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Found RTP audio format 8 [Oct 3 16:01:46] DEBUG[9957][C-00000001] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f3ef02b5470 [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Found RTP audio format 101 [Oct 3 16:01:46] DEBUG[9957][C-00000001] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f3ef02b5470 [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED. [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 3 16:01:46] DEBUG[9957][C-00000001] acl.c: For destination '10.10.10.100', our source address is '10.10.10.103'. [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Peer audio RTP is at port 10.10.10.100:7078 [Oct 3 16:01:46] DEBUG[9957][C-00000001] rtp_engine.c: Copying payload 8 (0x7f3f3000e698) from 0x7f3ef02b5470 to 0x7f3f200077d8 [Oct 3 16:01:46] DEBUG[9957][C-00000001] rtp_engine.c: Copying payload 101 (0x7f3f30002a48) from 0x7f3ef02b5470 to 0x7f3f200077d8 [Oct 3 16:01:46] DEBUG[9957][C-00000001] acl.c: Attached to given IP address [Oct 3 16:01:46] DEBUG[9957][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f3f20007610' [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: We're settling with these formats: (alaw) [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Setting native formats after processing SDP. peer joint formats (alaw), old nativeformats (alaw) [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Got a SIP re-invite for call 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: SIP/bob-00000003: This call is UP.... [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK1191253294;received=10.10.10.100;rport=5060 From: ;tag=1225084157 To: "alice" ;tag=as340f262a Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 3 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.10.10.100:5060 [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: ** Our native-bridge filtered capablity: (alaw) [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: ** Our prefcodec: (alaw) [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Audio is at 16778 [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Adding codec alaw to SDP [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Adding codec ulaw to SDP [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Adding codec gsm to SDP [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Setting framing on incoming call: 0 [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 3 16:01:46] VERBOSE[9957][C-00000001] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.10.10.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK1191253294;received=10.10.10.100;rport=5060 From: ;tag=1225084157 To: "alice" ;tag=as340f262a Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 3 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 310 v=0 o=root 1842940510 1842940513 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.100:5060 [Oct 3 16:01:46] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> jaK <-------------> [Oct 3 16:01:46] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Tech starting 'SIP/alice-00000002' and 'SIP/bob-00000003' with target 'SIP/alice-00000002' [Oct 3 16:01:46] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Sending 'SIP/alice-00000002' back to remote [Oct 3 16:01:46] DEBUG[10165][C-00000001] chan_sip.c: Sending reinvite on SIP 'kjERd3dZtA5kWmHI@10.10.10.2' - It's audio soon redirected to IP 10.10.10.100:7078 [Oct 3 16:01:46] DEBUG[10165][C-00000001] chan_sip.c: Strict routing enforced for session kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:46] VERBOSE[10165][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:46] VERBOSE[10165][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 3 16:01:46] DEBUG[10165][C-00000001] chan_sip.c: ** Our native-bridge filtered capablity: (alaw) [Oct 3 16:01:46] DEBUG[10165][C-00000001] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 3 16:01:46] DEBUG[10165][C-00000001] chan_sip.c: ** Our prefcodec: (nothing) [Oct 3 16:01:46] VERBOSE[10165][C-00000001] chan_sip.c: Audio is at 16192 [Oct 3 16:01:46] VERBOSE[10165][C-00000001] chan_sip.c: Adding codec alaw to SDP [Oct 3 16:01:46] VERBOSE[10165][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 3 16:01:46] DEBUG[10165][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Oct 3 16:01:46] DEBUG[10165][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 3 16:01:46] DEBUG[10165][C-00000001] chan_sip.c: Initializing already initialized SIP dialog kjERd3dZtA5kWmHI@10.10.10.2 (presumably reinvite) [Oct 3 16:01:46] VERBOSE[10165][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.2:5060: INVITE sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK539c1bcb Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 104 INVITE User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported 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: 263 v=0 o=root 216698452 216698455 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.100 t=0 0 m=audio 7078 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Oct 3 16:01:46] DEBUG[10165][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:46] DEBUG[10165][C-00000001] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 3 16:01:46] VERBOSE[10165][C-00000001] res_musiconhold.c: Stopped music on hold on SIP/alice-00000002 [Oct 3 16:01:46] DEBUG[10165][C-00000001] channel.c: Channel SIP/alice-00000002 setting write format path: alaw -> alaw [Oct 3 16:01:46] DEBUG[10165][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 3 16:01:46] DEBUG[10165][C-00000001] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 3 16:01:46] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Tech starting 'SIP/alice-00000002' and 'SIP/bob-00000003' with target 'SIP/alice-00000002' [Oct 3 16:01:46] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Sending 'SIP/alice-00000002' back to remote [Oct 3 16:01:46] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> ACK sip:alice@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport;branch=z9hG4bK29639177 From: ;tag=1225084157 To: "alice" ;tag=as340f262a Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 3 ACK Contact: Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 3 16:01:46] VERBOSE[9957] chan_sip.c: --- (10 headers 0 lines) --- [Oct 3 16:01:46] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on '6047941670ac6f605ace08a861ed2402@10.10.10.103:5060' of Response 3: Match Found [Oct 3 16:01:47] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK539c1bcb Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 104 INVITE From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Supported: replaces Content-Type: application/sdp Content-Length: 188 v=0 o=- 87976961 20420308 IN IP4 10.10.10.2 s=SIP CALL c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Oct 3 16:01:47] VERBOSE[9957] chan_sip.c: --- (11 headers 9 lines) --- [Oct 3 16:01:47] DEBUG[9957][C-00000001] chan_sip.c: Acked pending invite 104 [Oct 3 16:01:47] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on 'kjERd3dZtA5kWmHI@10.10.10.2' of Request 104: Match Found [Oct 3 16:01:47] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 3 16:01:47] DEBUG[9957][C-00000001] chan_sip.c: Call kjERd3dZtA5kWmHI@10.10.10.2 responded to our reinvite without changing SDP version; ignoring SDP. [Oct 3 16:01:47] DEBUG[9957][C-00000001] chan_sip.c: Updating call counter for incoming call [Oct 3 16:01:47] DEBUG[9957][C-00000001] chan_sip.c: Strict routing enforced for session kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:47] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:47] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 3 16:01:47] VERBOSE[9957][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.10.10.2:5060: ACK sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK15d3631a Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 104 ACK User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Content-Length: 0 --- [Oct 3 16:01:47] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'ACK sip:ali' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:47] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Tech starting 'SIP/alice-00000002' and 'SIP/bob-00000003' with target 'SIP/bob-00000003' [Oct 3 16:01:47] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Sending 'SIP/bob-00000003' back to remote [Oct 3 16:01:48] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.100:5060 ---> BYE sip:alice@10.10.10.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport;branch=z9hG4bK384149759 From: ;tag=1225084157 To: "alice" ;tag=as340f262a Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 4 BYE Contact: Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> [Oct 3 16:01:48] VERBOSE[9957] chan_sip.c: --- (10 headers 0 lines) --- [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Initializing initreq for method BYE - callid 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 [Oct 3 16:01:48] VERBOSE[9957][C-00000001] chan_sip.c: Sending to 10.10.10.100:5060 (no NAT) [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Setting SIP_ALREADYGONE on dialog 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 [Oct 3 16:01:48] VERBOSE[9957][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '6047941670ac6f605ace08a861ed2402@10.10.10.103:5060' in 6400 ms (Method: BYE) [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Received bye, issuing owner hangup [Oct 3 16:01:48] VERBOSE[9957][C-00000001] chan_sip.c: <--- Transmitting (no NAT) to 10.10.10.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK384149759;received=10.10.10.100;rport=5060 From: ;tag=1225084157 To: "alice" ;tag=as340f262a Call-ID: 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 CSeq: 4 BYE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.10.100:5060 [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge_channel.c: Setting 0x7f3f2001e540(SIP/bob-00000003) state from:0 to:1 [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge_channel.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: pulling 0x7f3f2001e540(SIP/bob-00000003) [Oct 3 16:01:48] VERBOSE[10166][C-00000001] bridge_channel.c: Channel SIP/bob-00000003 left 'native_rtp' basic-bridge <6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9> [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge_channel.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: 0x7f3f2001e540(SIP/bob-00000003) is leaving native_rtp technology [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Channel 'SIP/bob-00000003' is leaving bridge tech [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Detaching hook data 0x7f3f20018af8 from 'SIP/bob-00000003' [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Tech stopping 'SIP/alice-00000002' and 'SIP/bob-00000003' with target 'none' [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Bringing back 'SIP/alice-00000002' and 'SIP/bob-00000003' to us [Oct 3 16:01:48] DEBUG[10166][C-00000001] chan_sip.c: Sending reinvite on SIP 'kjERd3dZtA5kWmHI@10.10.10.2' - It's audio soon redirected to IP 10.10.10.103:5060 [Oct 3 16:01:48] DEBUG[10166][C-00000001] acl.c: Attached to given IP address [Oct 3 16:01:48] DEBUG[10166][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f3f3000b9b0' [Oct 3 16:01:48] DEBUG[10166][C-00000001] chan_sip.c: Strict routing enforced for session kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:48] VERBOSE[10166][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:48] VERBOSE[10166][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 3 16:01:48] DEBUG[10166][C-00000001] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True [Oct 3 16:01:48] DEBUG[10166][C-00000001] chan_sip.c: ** Our prefcodec: (nothing) [Oct 3 16:01:48] VERBOSE[10166][C-00000001] chan_sip.c: Audio is at 16192 [Oct 3 16:01:48] VERBOSE[10166][C-00000001] chan_sip.c: Adding codec alaw to SDP [Oct 3 16:01:48] VERBOSE[10166][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 3 16:01:48] DEBUG[10166][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Oct 3 16:01:48] DEBUG[10166][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (alaw) [Oct 3 16:01:48] DEBUG[10166][C-00000001] chan_sip.c: Initializing already initialized SIP dialog kjERd3dZtA5kWmHI@10.10.10.2 (presumably reinvite) [Oct 3 16:01:48] VERBOSE[10166][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.2:5060: INVITE sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK050d90e9 Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 105 INVITE User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported 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: 264 v=0 o=root 216698452 216698456 IN IP4 10.10.10.103 s=Asterisk PBX UNKNOWN__and_probably_unsupported c=IN IP4 10.10.10.103 t=0 0 m=audio 16192 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 3 16:01:48] DEBUG[10166][C-00000001] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/alice-00000002' and 'SIP/bob-00000003' - media will flow through Asterisk core [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge_native_rtp.c: Destroying channel tech_pvt data 0x7f3f2001bb80 [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: dissolving bridge with cause 16(Normal Clearing) [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge_channel.c: Setting 0x7f3f200072d0(SIP/alice-00000002) state from:0 to:2 [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: queueing action type:13 sub:1001 [Oct 3 16:01:48] DEBUG[10166][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9 is dissolved, not performing smart bridge operation. [Oct 3 16:01:48] DEBUG[10166][C-00000001] channel.c: Channel 0x7f3f20019630 'SIP/bob-00000003' hanging up. Refs: 2 [Oct 3 16:01:48] DEBUG[10166][C-00000001] chan_sip.c: Hangup call SIP/bob-00000003, SIP callid 6047941670ac6f605ace08a861ed2402@10.10.10.103:5060 [Oct 3 16:01:48] DEBUG[10166][C-00000001] channel.c: Channel 0x7f3f20019630 'SIP/bob-00000003' destroying [Oct 3 16:01:48] DEBUG[9927] chan_sip.c: Checking device state for peer bob [Oct 3 16:01:48] DEBUG[9927] devicestate.c: Changing state for SIP/bob - state 1 (Not in use) [Oct 3 16:01:48] DEBUG[9928] cdr.c: Finalized CDR for SIP/alice-00000002 - start 1507039297.379729 answer 1507039298.820629 end 1507039308.456968 dispo ANSWERED [Oct 3 16:01:48] DEBUG[9928] cdr.c: CDR for SIP/bob-00000003 is dialed and has no Party B; discarding [Oct 3 16:01:48] DEBUG[10165][C-00000001] bridge_channel.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: pulling 0x7f3f200072d0(SIP/alice-00000002) [Oct 3 16:01:48] VERBOSE[10165][C-00000001] bridge_channel.c: Channel SIP/alice-00000002 left 'native_rtp' basic-bridge <6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9> [Oct 3 16:01:48] DEBUG[10165][C-00000001] bridge_channel.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: 0x7f3f200072d0(SIP/alice-00000002) is leaving native_rtp technology [Oct 3 16:01:48] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Channel 'SIP/alice-00000002' is leaving bridge tech [Oct 3 16:01:48] DEBUG[10165][C-00000001] bridge_native_rtp.c: Bridge '6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9'. Detaching hook data 0x7f3f20018d78 from 'SIP/alice-00000002' [Oct 3 16:01:48] DEBUG[10165][C-00000001] bridge_native_rtp.c: Destroying channel tech_pvt data 0x7f3f20019310 [Oct 3 16:01:48] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9 is dissolved, not performing smart bridge operation. [Oct 3 16:01:48] DEBUG[10165][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1407087384 to 315415363 due to a source change [Oct 3 16:01:48] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: actually destroying basic bridge, nobody wants it anymore [Oct 3 16:01:48] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: calling basic bridge destructor [Oct 3 16:01:48] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: calling native_rtp technology stop [Oct 3 16:01:48] DEBUG[10165][C-00000001] bridge.c: Bridge 6b5b4823-1a72-43b0-bc5c-c77e6a97c0c9: calling native_rtp technology destructor [Oct 3 16:01:48] DEBUG[10165][C-00000001] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Oct 3 16:01:48] DEBUG[10165][C-00000001] pbx.c: Spawn extension (local,222,1) exited non-zero on 'SIP/alice-00000002' [Oct 3 16:01:48] VERBOSE[10165][C-00000001] pbx.c: Spawn extension (local, 222, 1) exited non-zero on 'SIP/alice-00000002' [Oct 3 16:01:48] DEBUG[10165][C-00000001] channel.c: Soft-Hanging (0x10) up channel 'SIP/alice-00000002' [Oct 3 16:01:48] DEBUG[10165][C-00000001] pbx_lua.c: Looking up h@local:1 [Oct 3 16:01:48] DEBUG[10165][C-00000001] pbx_lua.c: Looking up h@default:1 [Oct 3 16:01:48] DEBUG[10165][C-00000001] pbx_lua.c: Looking up h@demo:1 [Oct 3 16:01:48] DEBUG[10165][C-00000001] pbx_lua.c: Looking up h@demo:1 [Oct 3 16:01:48] DEBUG[10165][C-00000001] channel.c: Channel 0x7f3f30018f70 'SIP/alice-00000002' hanging up. Refs: 2 [Oct 3 16:01:48] DEBUG[10165][C-00000001] chan_sip.c: Hangup call SIP/alice-00000002, SIP callid kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:48] VERBOSE[10165][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog 'kjERd3dZtA5kWmHI@10.10.10.2' in 6400 ms (Method: ACK) [Oct 3 16:01:48] DEBUG[10165][C-00000001] channel.c: Channel 0x7f3f30018f70 'SIP/alice-00000002' destroying [Oct 3 16:01:48] DEBUG[9928] res_config_sqlite.c: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"alice" ','alice','222','local','SIP/alice-00000002','SIP/bob-00000003','Dial','SIP/bob','2017-10-03 16:01:37','2017-10-03 16:01:38','2017-10-03 16:01:48','11','9','ANSWERED','DOCUMENTATION','1507039297.3') [Oct 3 16:01:48] DEBUG[9927] chan_sip.c: Checking device state for peer alice [Oct 3 16:01:48] DEBUG[9927] devicestate.c: Changing state for SIP/alice - state 1 (Not in use) [Oct 3 16:01:48] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK050d90e9 Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 105 INVITE From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Supported: replaces Content-Type: application/sdp Content-Length: 188 v=0 o=- 66367565 29296070 IN IP4 10.10.10.2 s=SIP CALL c=IN IP4 10.10.10.2 t=0 0 m=audio 1780 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Oct 3 16:01:48] VERBOSE[9957] chan_sip.c: --- (11 headers 9 lines) --- [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Acked pending invite 105 [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on 'kjERd3dZtA5kWmHI@10.10.10.2' of Request 105: Match Found [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Call kjERd3dZtA5kWmHI@10.10.10.2 responded to our reinvite without changing SDP version; ignoring SDP. [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Updating call counter for incoming call [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Strict routing enforced for session kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:48] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:48] VERBOSE[9957][C-00000001] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 3 16:01:48] VERBOSE[9957][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.10.10.2:5060: ACK sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK15c74b6c Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 105 ACK User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Content-Length: 0 --- [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Trying to put 'ACK sip:ali' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:48] DEBUG[9957] chan_sip.c: Strict routing enforced for session kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:48] VERBOSE[9957] chan_sip.c: set_destination: Parsing for address/port to send to [Oct 3 16:01:48] VERBOSE[9957] chan_sip.c: set_destination: set destination to 10.10.10.2:5060 [Oct 3 16:01:48] VERBOSE[9957] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.10.2:5060: BYE sip:alice@10.10.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK469589b6 Max-Forwards: 70 From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 106 BYE User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported Proxy-Authorization: Digest username="alice", realm="asterisk", algorithm=MD5, uri="sip:10.10.10.103", nonce="3986e382", response="fd1e06b4b087e9ae412e80c7f9d89119" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Oct 3 16:01:48] DEBUG[9957] chan_sip.c: Trying to put 'BYE sip:ali' onto UDP socket destined for 10.10.10.2:5060 [Oct 3 16:01:48] VERBOSE[9957] chan_sip.c: Scheduling destruction of SIP dialog 'kjERd3dZtA5kWmHI@10.10.10.2' in 6400 ms (Method: ACK) [Oct 3 16:01:48] VERBOSE[9957] chan_sip.c: <--- SIP read from UDP:10.10.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.103:5060;branch=z9hG4bK469589b6 Call-ID: kjERd3dZtA5kWmHI@10.10.10.2 CSeq: 106 BYE From: "222" ;tag=as237823d9 To: "alice" ;tag=pnZtIYPzVusub6w1 Contact: Content-Length: 0 <-------------> [Oct 3 16:01:48] VERBOSE[9957] chan_sip.c: --- (8 headers 0 lines) --- [Oct 3 16:01:48] DEBUG[9957][C-00000001] chan_sip.c: Stopping retransmission on 'kjERd3dZtA5kWmHI@10.10.10.2' of Request 106: Match Found [Oct 3 16:01:48] DEBUG[9957] chan_sip.c: Destroying SIP dialog kjERd3dZtA5kWmHI@10.10.10.2 [Oct 3 16:01:48] VERBOSE[9957] chan_sip.c: Really destroying SIP dialog 'kjERd3dZtA5kWmHI@10.10.10.2' Method: ACK [Oct 3 16:01:48] DEBUG[9957] rtp_engine.c: Destroyed RTP instance '0x7f3f3000b9b0'