[Jul 14 11:56:47] VERBOSE[31008] chan_sip.c: <--- SIP read from UDP:10.34.2.106:5060 ---> INVITE sip:112@10.34.2.12;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.34.2.106:5060;branch=z9hG4bK57948bf5dd69bc001 Max-Forwards: 70 From: "Alice" ;tag=7cacdf11b1 To: Call-ID: 93d1bec9bae4952e CSeq: 18101 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Alice" ;+sip.instance="" Supported: path, gruu, 100rel, replaces User-Agent: Aastra 6731i/3.3.1.4295 Content-Type: application/sdp Content-Length: 302 v=0 o=MxSIP 0 1 IN IP4 10.34.2.106 s=SIP Call c=IN IP4 10.34.2.106 t=0 0 m=audio 3000 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jul 14 11:56:47] VERBOSE[31008] chan_sip.c: --- (14 headers 15 lines) --- [Jul 14 11:56:47] VERBOSE[31008] chan_sip.c: Sending to 10.34.2.106:5060 (no NAT) [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Sending to 10.34.2.106:5060 (no NAT) [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Using INVITE request as basis request - 93d1bec9bae4952e [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Found peer 'alice' for 'alice' from 10.34.2.106:5060 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.34.2.106:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.34.2.106:5060;branch=z9hG4bK57948bf5dd69bc001;received=10.34.2.106 From: "Alice" ;tag=7cacdf11b1 To: ;tag=as177f7408 Call-ID: 93d1bec9bae4952e CSeq: 18101 INVITE Server: Asterisk PBX 13.4.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7204c808" Content-Length: 0 <------------> [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Scheduling destruction of SIP dialog '93d1bec9bae4952e' in 32000 ms (Method: INVITE) [Jul 14 11:56:47] VERBOSE[31008] chan_sip.c: <--- SIP read from UDP:10.34.2.106:5060 ---> ACK sip:112@10.34.2.12;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.34.2.106:5060;branch=z9hG4bK57948bf5dd69bc001 Max-Forwards: 70 From: "Alice" ;tag=7cacdf11b1 To: ;tag=as177f7408 Call-ID: 93d1bec9bae4952e CSeq: 18101 ACK User-Agent: Aastra 6731i/3.3.1.4295 Content-Length: 0 <-------------> [Jul 14 11:56:47] VERBOSE[31008] chan_sip.c: --- (9 headers 0 lines) --- [Jul 14 11:56:47] VERBOSE[31008] chan_sip.c: <--- SIP read from UDP:10.34.2.106:5060 ---> INVITE sip:112@10.34.2.12;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.34.2.106:5060;branch=z9hG4bKed130abd7f45c8dfb Max-Forwards: 70 From: "Alice" ;tag=7cacdf11b1 To: Call-ID: 93d1bec9bae4952e CSeq: 18102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="alice",realm="asterisk",nonce="7204c808",uri="sip:112@10.34.2.12;user=phone",response="00301b9d404fe690ae99bb0e634e6486",algorithm=MD5 Contact: "Alice" ;+sip.instance="" Supported: path, gruu, 100rel, replaces User-Agent: Aastra 6731i/3.3.1.4295 Content-Type: application/sdp Content-Length: 302 v=0 o=MxSIP 0 1 IN IP4 10.34.2.106 s=SIP Call c=IN IP4 10.34.2.106 t=0 0 m=audio 3000 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jul 14 11:56:47] VERBOSE[31008] chan_sip.c: --- (15 headers 15 lines) --- [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Sending to 10.34.2.106:5060 (no NAT) [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Using INVITE request as basis request - 93d1bec9bae4952e [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Found peer 'alice' for 'alice' from 10.34.2.106:5060 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] netsock2.c: Using SIP RTP CoS mark 5 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Found RTP audio format 8 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Found RTP audio format 0 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Found RTP audio format 18 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Found RTP audio format 101 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Found audio description format PCMA for ID 8 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Found audio description format G729 for ID 18 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Peer audio RTP is at port 10.34.2.106:3000 [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: Looking for 112 in internal (domain 10.34.2.12) [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] sip/route.c: sip_route_dump: route/path hop: [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] chan_sip.c: <--- Transmitting (no NAT) to 10.34.2.106:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.34.2.106:5060;branch=z9hG4bKed130abd7f45c8dfb;received=10.34.2.106 From: "Alice" ;tag=7cacdf11b1 To: Call-ID: 93d1bec9bae4952e CSeq: 18102 INVITE Server: Asterisk PBX 13.4.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] pbx.c: Executing [112@internal:1] NoOp("SIP/alice-00000035", "") in new stack [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] pbx.c: Executing [112@internal:2] Dial("SIP/alice-00000035", "Local/102@internal") in new stack [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] app_dial.c: Called Local/102@internal [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] pbx.c: Executing [102@internal:1] NoOp("Local/102@internal-00000012;2", "") in new stack [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] pbx.c: Executing [102@internal:2] Answer("Local/102@internal-00000012;2", "") in new stack [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] app_dial.c: Local/102@internal-00000012;1 answered SIP/alice-00000035 [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] chan_sip.c: Audio is at 15672 [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] chan_sip.c: Adding codec ulaw to SDP [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] chan_sip.c: Adding codec alaw to SDP [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] chan_sip.c: Adding codec gsm to SDP [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.34.2.106:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.34.2.106:5060;branch=z9hG4bKed130abd7f45c8dfb;received=10.34.2.106 From: "Alice" ;tag=7cacdf11b1 To: ;tag=as6ae26eea Call-ID: 93d1bec9bae4952e CSeq: 18102 INVITE Server: Asterisk PBX 13.4.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 282 v=0 o=root 1693605491 1693605491 IN IP4 10.34.2.12 s=Asterisk PBX 13.4.0 c=IN IP4 10.34.2.12 t=0 0 m=audio 15672 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv <------------> [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] bridge_channel.c: Channel SIP/alice-00000035 joined 'simple_bridge' basic-bridge <745d72f9-a379-43a9-9359-3314971adeff> [Jul 14 11:56:47] VERBOSE[13955][C-0000001b] bridge_channel.c: Channel Local/102@internal-00000012;1 joined 'simple_bridge' basic-bridge <745d72f9-a379-43a9-9359-3314971adeff> [Jul 14 11:56:47] VERBOSE[31008] chan_sip.c: <--- SIP read from UDP:10.34.2.106:5060 ---> ACK sip:112@10.34.2.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.34.2.106:5060;branch=z9hG4bK762113cc626b17bb0 Max-Forwards: 70 From: "Alice" ;tag=7cacdf11b1 To: ;tag=as6ae26eea Call-ID: 93d1bec9bae4952e CSeq: 18102 ACK Authorization: Digest username="alice",realm="asterisk",nonce="7204c808",uri="sip:112@10.34.2.12;user=phone",response="00301b9d404fe690ae99bb0e634e6486",algorithm=MD5 User-Agent: Aastra 6731i/3.3.1.4295 Content-Length: 0 <-------------> [Jul 14 11:56:47] VERBOSE[31008] chan_sip.c: --- (10 headers 0 lines) --- [Jul 14 11:56:47] VERBOSE[13953][C-0000001b] res_rtp_asterisk.c: 0xa3471c0 -- Probation passed - setting RTP source address to 10.34.2.106:3000 [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] pbx.c: Executing [102@internal:3] Dial("Local/102@internal-00000012;2", "SIP/bob") in new stack [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] netsock2.c: Using SIP RTP CoS mark 5 [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] chan_sip.c: Audio is at 19266 [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] chan_sip.c: Adding codec ulaw to SDP [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] chan_sip.c: Adding codec alaw to SDP [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] chan_sip.c: Adding codec gsm to SDP [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] chan_sip.c: Reliably Transmitting (no NAT) to 10.34.2.101:5060: INVITE sip:bob@10.34.2.101:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.34.2.12:5060;branch=z9hG4bK64313d67 Max-Forwards: 70 From: "Alice" ;tag=as52ffafbc To: Contact: Call-ID: 705e6a58285ff5ec5e580a9a1703dc86@10.34.2.12:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.4.0 Date: Tue, 14 Jul 2015 15:56:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 278 v=0 o=root 20551888 20551888 IN IP4 10.34.2.12 s=Asterisk PBX 13.4.0 c=IN IP4 10.34.2.12 t=0 0 m=audio 19266 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] app_dial.c: Called SIP/bob [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] app_dial.c: Local/102@internal-00000012;2 requested media update control 26, passing it to SIP/bob-00000036 [Jul 14 11:56:47] VERBOSE[31008] chan_sip.c: <--- SIP read from UDP:10.34.2.101:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.34.2.12:5060;branch=z9hG4bK64313d67 From: "Alice" ;tag=as52ffafbc To: ;tag=3685690702 Call-ID: 705e6a58285ff5ec5e580a9a1703dc86@10.34.2.12:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Bob" ;+sip.instance="" Server: Aastra 6731i/3.3.1.4295 Supported: path Content-Length: 0 <-------------> [Jul 14 11:56:47] VERBOSE[31008] chan_sip.c: --- (12 headers 0 lines) --- [Jul 14 11:56:47] VERBOSE[31008][C-0000001b] sip/route.c: sip_route_dump: route/path hop: [Jul 14 11:56:47] VERBOSE[13954][C-0000001b] app_dial.c: SIP/bob-00000036 is ringing [Jul 14 11:56:49] VERBOSE[31008] chan_sip.c: <--- SIP read from UDP:10.34.2.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.34.2.12:5060;branch=z9hG4bK64313d67 From: "Alice" ;tag=as52ffafbc To: ;tag=3685690702 Call-ID: 705e6a58285ff5ec5e580a9a1703dc86@10.34.2.12:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Bob" ;+sip.instance="" Server: Aastra 6731i/3.3.1.4295 Supported: path, replaces Content-Type: application/sdp Content-Length: 243 v=0 o=MxSIP 0 1 IN IP4 10.34.2.101 s=SIP Call c=IN IP4 10.34.2.101 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=sendrecv <-------------> [Jul 14 11:56:49] VERBOSE[31008] chan_sip.c: --- (13 headers 12 lines) --- [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: Found RTP audio format 0 [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: Found RTP audio format 8 [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: Found RTP audio format 101 [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: Found audio description format PCMA for ID 8 [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: Peer audio RTP is at port 10.34.2.101:3000 [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] sip/route.c: sip_route_dump: route/path hop: [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: set_destination: set destination to 10.34.2.101:5060 [Jul 14 11:56:49] VERBOSE[31008][C-0000001b] chan_sip.c: Transmitting (no NAT) to 10.34.2.101:5060: ACK sip:bob@10.34.2.101:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.34.2.12:5060;branch=z9hG4bK66493bc5 Max-Forwards: 70 From: "Alice" ;tag=as52ffafbc To: ;tag=3685690702 Contact: Call-ID: 705e6a58285ff5ec5e580a9a1703dc86@10.34.2.12:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.4.0 Content-Length: 0 --- [Jul 14 11:56:49] VERBOSE[13954][C-0000001b] app_dial.c: SIP/bob-00000036 answered Local/102@internal-00000012;2 [Jul 14 11:56:49] VERBOSE[13954][C-0000001b] bridge_channel.c: Channel Local/102@internal-00000012;2 joined 'simple_bridge' basic-bridge [Jul 14 11:56:49] VERBOSE[13958][C-0000001b] bridge_channel.c: Channel SIP/bob-00000036 joined 'simple_bridge' basic-bridge [Jul 14 11:56:49] VERBOSE[13955][C-0000001b] bridge.c: Move-swap optimizing Local/102@internal-00000012;2 <-- SIP/alice-00000035. [Jul 14 11:56:49] VERBOSE[13955][C-0000001b] bridge_channel.c: Channel SIP/alice-00000035 left 'simple_bridge' basic-bridge <745d72f9-a379-43a9-9359-3314971adeff> [Jul 14 11:56:49] VERBOSE[13955][C-0000001b] bridge_channel.c: Channel Local/102@internal-00000012;2 left 'simple_bridge' basic-bridge [Jul 14 11:56:49] VERBOSE[13955][C-0000001b] bridge_channel.c: Channel SIP/alice-00000035 swapped with Local/102@internal-00000012;2 into 'simple_bridge' basic-bridge [Jul 14 11:56:49] VERBOSE[13955][C-0000001b] bridge.c: Bridge abc05116-c51c-4ec6-9a2e-1d0d14a61b44: switching from simple_bridge technology to native_rtp [Jul 14 11:56:49] VERBOSE[13955][C-0000001b] bridge_native_rtp.c: Locally RTP bridged 'SIP/alice-00000035' and 'SIP/bob-00000036' in stack [Jul 14 11:56:49] VERBOSE[13955][C-0000001b] bridge_native_rtp.c: Locally RTP bridged 'SIP/alice-00000035' and 'SIP/bob-00000036' in stack [Jul 14 11:56:49] VERBOSE[13955][C-0000001b] bridge_channel.c: Channel Local/102@internal-00000012;1 left 'simple_bridge' basic-bridge <745d72f9-a379-43a9-9359-3314971adeff> [Jul 14 11:56:49] VERBOSE[13954][C-0000001b] pbx.c: Spawn extension (internal, 102, 3) exited non-zero on 'Local/102@internal-00000012;2' [Jul 14 11:56:50] VERBOSE[13958][C-0000001b] res_rtp_asterisk.c: 0x9c67d90 -- Probation passed - setting RTP source address to 10.34.2.101:3000