[2015-03-27 09:46:32] Asterisk SVN-branch-13-r433338M built by root @ 172.17.3.1 on a i686 running Linux on 2014-11-08 11:14:46 UTC [2015-03-27 09:46:31] VERBOSE[16193] config.c: Parsing '/etc/asterisk/logger.conf': Found [2015-03-27 09:46:32] VERBOSE[16193] logger.c: Asterisk Queue Logger restarted [2015-03-27 09:46:43] DEBUG[15866] chan_iax2.c: ip callno count decremented to 2 for 85.25.208.239 [2015-03-27 09:46:43] DEBUG[15876] chan_iax2.c: ip callno count incremented to 3 for 85.25.208.239 [2015-03-27 09:46:43] DEBUG[15877] chan_iax2.c: schedule decrement of callno used for 85.25.208.239 in 60 seconds [2015-03-27 09:46:43] DEBUG[15877] chan_iax2.c: Peer 12346178: got pong, lastms 27, historicms 27, maxms 2000 [2015-03-27 09:46:45] DEBUG[15866] chan_iax2.c: ip callno count decremented to 2 for 85.25.208.239 [2015-03-27 09:46:47] DEBUG[15866] chan_iax2.c: ip callno count decremented to 7 for 92.226.232.223 [2015-03-27 09:46:47] DEBUG[15866] chan_iax2.c: ip callno count decremented to 6 for 92.226.232.223 [2015-03-27 09:46:47] DEBUG[15866] chan_iax2.c: ip callno count decremented to 3 for 77.37.11.180 [2015-03-27 09:46:47] DEBUG[15870] chan_iax2.c: schedule decrement of callno used for 92.226.232.223 in 60 seconds [2015-03-27 09:46:47] DEBUG[15872] chan_iax2.c: schedule decrement of callno used for 92.226.232.223 in 60 seconds [2015-03-27 09:46:47] DEBUG[15874] chan_iax2.c: schedule decrement of callno used for 77.37.11.180 in 60 seconds [2015-03-27 09:46:55] DEBUG[15891] pbx_dundi.c: Register us as '00:40:63:e3:96:05' to '00:e0:4d:6c:93:72' [2015-03-27 09:46:57] DEBUG[15868] chan_iax2.c: ip callno count incremented to 7 for 92.226.232.223 [2015-03-27 09:46:57] DEBUG[15869] chan_iax2.c: ip callno count incremented to 8 for 92.226.232.223 [2015-03-27 09:46:57] DEBUG[15870] chan_iax2.c: ip callno count incremented to 4 for 77.37.11.180 [2015-03-27 09:46:58] DEBUG[16194] threadpool.c: Worker thread idle timeout reached. Dying. [2015-03-27 09:46:58] DEBUG[15785] threadpool.c: Destroying worker thread 13 [2015-03-27 09:47:00] NOTICE[15891] pbx_dundi.c: Max retries exceeded to host '172.17.1.101:4520' msg 0 on call 8974 [2015-03-27 09:47:07] DEBUG[15866] chan_iax2.c: ip callno count decremented to 7 for 92.226.232.223 [2015-03-27 09:47:07] DEBUG[15866] chan_iax2.c: ip callno count decremented to 6 for 92.226.232.223 [2015-03-27 09:47:07] DEBUG[15866] chan_iax2.c: ip callno count decremented to 3 for 77.37.11.180 [2015-03-27 09:47:07] DEBUG[15875] chan_iax2.c: schedule decrement of callno used for 92.226.232.223 in 60 seconds [2015-03-27 09:47:07] DEBUG[15877] chan_iax2.c: schedule decrement of callno used for 92.226.232.223 in 60 seconds [2015-03-27 09:47:07] DEBUG[15869] chan_iax2.c: schedule decrement of callno used for 77.37.11.180 in 60 seconds [2015-03-27 09:47:10] DEBUG[15861] chan_sip.c: Auto destroying SIP dialog '19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB' [2015-03-27 09:47:10] DEBUG[15861] chan_sip.c: Destroying SIP dialog 19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB [2015-03-27 09:47:10] VERBOSE[15861] chan_sip.c: Really destroying SIP dialog '19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB' Method: REGISTER [2015-03-27 09:47:17] DEBUG[15873] chan_iax2.c: ip callno count incremented to 7 for 92.226.232.223 [2015-03-27 09:47:17] DEBUG[15874] chan_iax2.c: ip callno count incremented to 8 for 92.226.232.223 [2015-03-27 09:47:17] DEBUG[15875] chan_iax2.c: ip callno count incremented to 4 for 77.37.11.180 [2015-03-27 09:47:20] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> REGISTER sip:172.17.3.1 SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjKkPW3V3ZEhBJlWYyv-t2sjxAMEUy4JGE;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=zZY5VTTTsOF-cY4PRP1p076c5wnKzYo2 To: "SIP Test Client" Call-ID: 19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB CSeq: 50213 REGISTER Authorization: Digest username="testClient", realm="asterisk", nonce="72282c02", uri="sip:172.17.3.1", response="706dd2a2936d4a0cfbb5ba53f2437bef", algorithm=MD5 User-Agent: Bria iOS 3.3.0 Supported: outbound, path Contact: "SIP Test Client" ;reg-id=1;+sip.instance="" Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 <-------------> [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 0 [ 42]: REGISTER sip:172.17.3.1 SIP/2.0 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjKkPW3V3ZEhBJlWYyv-t2sjxAMEUy4JGE;alias [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=zZY5VTTTsOF-cY4PRP1p076c5wnKzYo2 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 4 [ 51]: To: "SIP Test Client" [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 5 [ 41]: Call-ID: 19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 6 [ 20]: CSeq: 50213 REGISTER [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 7 [170]: Authorization: Digest username="testClient", realm="asterisk", nonce="72282c02", uri="sip:172.17.3.1", response="706dd2a2936d4a0cfbb5ba53f2437bef", algorithm=MD5 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 8 [ 26]: User-Agent: Bria iOS 3.3.0 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 9 [ 25]: Supported: outbound, path [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 10 [147]: Contact: "SIP Test Client" ;reg-id=1;+sip.instance="" [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 11 [ 12]: Expires: 900 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 12 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [2015-03-27 09:47:20] VERBOSE[15883] chan_sip.c: --- (14 headers 0 lines) --- [2015-03-27 09:47:20] DEBUG[15883] acl.c: For destination '172.17.1.166', our source address is '172.17.3.1'. [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Setting AST_TRANSPORT_TLS with address 172.17.3.1:5061 [2015-03-27 09:47:20] VERBOSE[15883] chan_sip.c: Sending to 172.17.1.166:50199 (no NAT) [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Allocating new SIP dialog for 19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB - REGISTER (No RTP) [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Initializing initreq for method REGISTER - callid 19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB [2015-03-27 09:47:20] VERBOSE[15883] chan_sip.c: Sending to 172.17.1.166:50199 (no NAT) [2015-03-27 09:47:20] VERBOSE[15883] chan_sip.c: <--- Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPjKkPW3V3ZEhBJlWYyv-t2sjxAMEUy4JGE;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=zZY5VTTTsOF-cY4PRP1p076c5wnKzYo2 To: "SIP Test Client" ;tag=as5d7f675b Call-ID: 19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB CSeq: 50213 REGISTER Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c42c4eb" Content-Length: 0 <------------> [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Trying to put 'SIP/2.0 401' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:20] VERBOSE[15883] chan_sip.c: Scheduling destruction of SIP dialog '19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB' in 32000 ms (Method: REGISTER) [2015-03-27 09:47:20] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> REGISTER sip:172.17.3.1 SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjoxbMGBhpogGW.PdwpBfujAlwz1U.3848;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=zZY5VTTTsOF-cY4PRP1p076c5wnKzYo2 To: "SIP Test Client" Call-ID: 19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB CSeq: 50214 REGISTER User-Agent: Bria iOS 3.3.0 Supported: outbound, path Contact: "SIP Test Client" ;reg-id=1;+sip.instance="" Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="testClient", realm="asterisk", nonce="6c42c4eb", uri="sip:172.17.3.1", response="b49a72d2b2fe8f613574e5ad7b49b78b", algorithm=MD5 Content-Length: 0 <-------------> [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 0 [ 42]: REGISTER sip:172.17.3.1 SIP/2.0 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjoxbMGBhpogGW.PdwpBfujAlwz1U.3848;alias [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=zZY5VTTTsOF-cY4PRP1p076c5wnKzYo2 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 4 [ 51]: To: "SIP Test Client" [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 5 [ 41]: Call-ID: 19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 6 [ 20]: CSeq: 50214 REGISTER [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 7 [ 26]: User-Agent: Bria iOS 3.3.0 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 8 [ 25]: Supported: outbound, path [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 9 [147]: Contact: "SIP Test Client" ;reg-id=1;+sip.instance="" [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 10 [ 12]: Expires: 900 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 11 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 12 [170]: Authorization: Digest username="testClient", realm="asterisk", nonce="6c42c4eb", uri="sip:172.17.3.1", response="b49a72d2b2fe8f613574e5ad7b49b78b", algorithm=MD5 [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [2015-03-27 09:47:20] VERBOSE[15883] chan_sip.c: --- (14 headers 0 lines) --- [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Initializing initreq for method REGISTER - callid 19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB [2015-03-27 09:47:20] VERBOSE[15883] chan_sip.c: Sending to 172.17.1.166:50199 (no NAT) [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Store REGISTER's src-IP:port for call routing. [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: build_path: do not use Path headers [2015-03-27 09:47:20] DEBUG[15785] threadpool.c: Increasing threadpool stasis-core's size by 1 [2015-03-27 09:47:20] VERBOSE[15883] chan_sip.c: <--- Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPjoxbMGBhpogGW.PdwpBfujAlwz1U.3848;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=zZY5VTTTsOF-cY4PRP1p076c5wnKzYo2 To: "SIP Test Client" ;tag=as5d7f675b Call-ID: 19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB CSeq: 50214 REGISTER Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 900 Contact: ;expires=900 Date: Fri, 27 Mar 2015 08:47:20 GMT Content-Length: 0 <------------> [2015-03-27 09:47:20] DEBUG[15883] chan_sip.c: Trying to put 'SIP/2.0 200' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:20] DEBUG[15797] devicestate.c: No provider found, checking channel drivers for SIP - testClient [2015-03-27 09:47:20] DEBUG[15797] chan_sip.c: Checking device state for peer testClient [2015-03-27 09:47:20] DEBUG[15797] devicestate.c: Changing state for SIP/testClient - state 1 (Not in use) [2015-03-27 09:47:20] VERBOSE[15883] chan_sip.c: Scheduling destruction of SIP dialog '19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB' in 32000 ms (Method: REGISTER) [2015-03-27 09:47:25] DEBUG[15869] chan_iax2.c: Allocate call number [2015-03-27 09:47:25] DEBUG[15869] chan_iax2.c: ip callno count incremented to 3 for 85.25.208.239 [2015-03-27 09:47:25] DEBUG[15869] chan_iax2.c: Registration created on call 9469 [2015-03-27 09:47:25] DEBUG[15874] chan_iax2.c: schedule decrement of callno used for 85.25.208.239 in 60 seconds [2015-03-27 09:47:26] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> INVITE sip:8501@172.17.3.1 SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjglKGq4p7NHafA0UJoah.i3JZMaitfXbs;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: Contact: "SIP Test Client" Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2644 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria iOS 3.3.0 Content-Type: application/sdp Content-Length: 1102 v=0 o=- 3636434846 3636434846 IN IP4 172.17.1.166 s=cpc_med c=IN IP4 172.17.1.166 b=AS:554 t=0 0 m=audio 4000 RTP/SAVP 111 110 109 103 9 0 8 18 101 a=sendrecv a=rtpmap:111 OPUS/48000/2 a=fmtp:111 maxplaybackrate=32000;useinbandfec=1 a=rtpmap:110 SILK/24000 a=fmtp:110 useinbandfec=1 a=rtpmap:109 SILK/16000 a=fmtp:109 useinbandfec=1 a=rtpmap:103 AMR-WB/16000 a=fmtp:103 octet-align=0 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7VPWc8um/bLNW2iWFR6tD6hciOqnvOYIQa9uMKYr a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:hNK5niRo8k8/Xbscu55VYsz7WPfg46mAtpRqR3L2 m=video 4002 RTP/SAVP 97 102 c=IN IP4 172.17.1.166 b=TIAS:512000 a=sendrecv a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42800c a=rtpmap:102 VP8/90000 a=orient:portrait a=rtcp-fb:* nack pli a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XmVadUHDw4fwentcubE+/bJ6DJis3xFXkLgBpFCs a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NDyHBo/dnJ8wUf4YDqOPZ/rWf+8zK/Ueyv7178Fk <-------------> [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 0 [ 45]: INVITE sip:8501@172.17.3.1 SIP/2.0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjglKGq4p7NHafA0UJoah.i3JZMaitfXbs;alias [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 4 [ 36]: To: [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 5 [ 70]: Contact: "SIP Test Client" [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 6 [ 41]: Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 7 [ 17]: CSeq: 2644 INVITE [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 8 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 9 [ 39]: Supported: replaces, 100rel, norefersub [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 10 [ 26]: User-Agent: Bria iOS 3.3.0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 12 [ 20]: Content-Length: 1102 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 13 [ 0]: [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 0 [ 3]: v=0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 1 [ 45]: o=- 3636434846 3636434846 IN IP4 172.17.1.166 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 2 [ 9]: s=cpc_med [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 3 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 4 [ 8]: b=AS:554 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 5 [ 5]: t=0 0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 6 [ 50]: m=audio 4000 RTP/SAVP 111 110 109 103 9 0 8 18 101 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 7 [ 10]: a=sendrecv [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 8 [ 25]: a=rtpmap:111 OPUS/48000/2 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 9 [ 47]: a=fmtp:111 maxplaybackrate=32000;useinbandfec=1 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 10 [ 23]: a=rtpmap:110 SILK/24000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 11 [ 25]: a=fmtp:110 useinbandfec=1 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 12 [ 23]: a=rtpmap:109 SILK/16000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 13 [ 25]: a=fmtp:109 useinbandfec=1 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 14 [ 25]: a=rtpmap:103 AMR-WB/16000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 15 [ 24]: a=fmtp:103 octet-align=0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 16 [ 20]: a=rtpmap:9 G722/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 17 [ 20]: a=rtpmap:0 PCMU/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 18 [ 20]: a=rtpmap:8 PCMA/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 19 [ 21]: a=rtpmap:18 G729/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 20 [ 19]: a=fmtp:18 annexb=no [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 21 [ 33]: a=rtpmap:101 telephone-event/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 22 [ 15]: a=fmtp:101 0-16 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 23 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7VPWc8um/bLNW2iWFR6tD6hciOqnvOYIQa9uMKYr [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 24 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:hNK5niRo8k8/Xbscu55VYsz7WPfg46mAtpRqR3L2 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 25 [ 28]: m=video 4002 RTP/SAVP 97 102 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 26 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 27 [ 13]: b=TIAS:512000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 28 [ 10]: a=sendrecv [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 29 [ 22]: a=rtpmap:97 H264/90000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 30 [ 33]: a=fmtp:97 profile-level-id=42800c [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 31 [ 22]: a=rtpmap:102 VP8/90000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 32 [ 17]: a=orient:portrait [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 33 [ 20]: a=rtcp-fb:* nack pli [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 34 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XmVadUHDw4fwentcubE+/bJ6DJis3xFXkLgBpFCs [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 35 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NDyHBo/dnJ8wUf4YDqOPZ/rWf+8zK/Ueyv7178Fk [2015-03-27 09:47:26] VERBOSE[15883] chan_sip.c: --- (13 headers 36 lines) --- [2015-03-27 09:47:26] DEBUG[15883] acl.c: For destination '172.17.1.166', our source address is '172.17.3.1'. [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Setting AST_TRANSPORT_TLS with address 172.17.3.1:5061 [2015-03-27 09:47:26] VERBOSE[15883] chan_sip.c: Sending to 172.17.1.166:50199 (no NAT) [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Allocating new SIP dialog for 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q - INVITE (No RTP) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, norefersub" [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sip/reqresp_parser.c: Found SIP option: -replaces- [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sip/reqresp_parser.c: Matched SIP option: replaces [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sip/reqresp_parser.c: Found SIP option: -100rel- [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sip/reqresp_parser.c: Matched SIP option: 100rel [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sip/reqresp_parser.c: Found SIP option: -norefersub- [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sip/reqresp_parser.c: Matched SIP option: norefersub [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Sending to 172.17.1.166:50199 (no NAT) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Using INVITE request as basis request - 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found peer 'testClient' for 'testClient' from 172.17.1.166:50199 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: <--- Reliably Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPjglKGq4p7NHafA0UJoah.i3JZMaitfXbs;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: ;tag=as3d53a939 Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2644 INVITE Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f5b9e6c" Content-Length: 0 <------------> [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 401' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q' in 32000 ms (Method: INVITE) [2015-03-27 09:47:26] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> ACK sip:8501@172.17.3.1 SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjglKGq4p7NHafA0UJoah.i3JZMaitfXbs;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: ;tag=as3d53a939 Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2644 ACK Content-Length: 0 <-------------> [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 0 [ 42]: ACK sip:8501@172.17.3.1 SIP/2.0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjglKGq4p7NHafA0UJoah.i3JZMaitfXbs;alias [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 4 [ 51]: To: ;tag=as3d53a939 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 5 [ 41]: Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 6 [ 14]: CSeq: 2644 ACK [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2015-03-27 09:47:26] VERBOSE[15883] chan_sip.c: --- (8 headers 0 lines) --- [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Stopping retransmission on '7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q' of Response 2644: Match Not Found [2015-03-27 09:47:26] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> INVITE sip:8501@172.17.3.1 SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjlu5Evd-yK18umT6bsgRcpGSzxmHAXd.S;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: Contact: "SIP Test Client" Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2645 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria iOS 3.3.0 Authorization: Digest username="testClient", realm="asterisk", nonce="6f5b9e6c", uri="sip:8501@172.17.3.1", response="ba654b95ac5ef5f0f6cdf0266df31b1b", algorithm=MD5 Content-Type: application/sdp Content-Length: 1102 v=0 o=- 3636434846 3636434846 IN IP4 172.17.1.166 s=cpc_med c=IN IP4 172.17.1.166 b=AS:554 t=0 0 m=audio 4000 RTP/SAVP 111 110 109 103 9 0 8 18 101 a=sendrecv a=rtpmap:111 OPUS/48000/2 a=fmtp:111 maxplaybackrate=32000;useinbandfec=1 a=rtpmap:110 SILK/24000 a=fmtp:110 useinbandfec=1 a=rtpmap:109 SILK/16000 a=fmtp:109 useinbandfec=1 a=rtpmap:103 AMR-WB/16000 a=fmtp:103 octet-align=0 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7VPWc8um/bLNW2iWFR6tD6hciOqnvOYIQa9uMKYr a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:hNK5niRo8k8/Xbscu55VYsz7WPfg46mAtpRqR3L2 m=video 4002 RTP/SAVP 97 102 c=IN IP4 172.17.1.166 b=TIAS:512000 a=sendrecv a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42800c a=rtpmap:102 VP8/90000 a=orient:portrait a=rtcp-fb:* nack pli a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XmVadUHDw4fwentcubE+/bJ6DJis3xFXkLgBpFCs a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NDyHBo/dnJ8wUf4YDqOPZ/rWf+8zK/Ueyv7178Fk <-------------> [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 0 [ 45]: INVITE sip:8501@172.17.3.1 SIP/2.0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjlu5Evd-yK18umT6bsgRcpGSzxmHAXd.S;alias [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 4 [ 36]: To: [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 5 [ 70]: Contact: "SIP Test Client" [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 6 [ 41]: Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 7 [ 17]: CSeq: 2645 INVITE [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 8 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 9 [ 39]: Supported: replaces, 100rel, norefersub [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 10 [ 26]: User-Agent: Bria iOS 3.3.0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 11 [175]: Authorization: Digest username="testClient", realm="asterisk", nonce="6f5b9e6c", uri="sip:8501@172.17.3.1", response="ba654b95ac5ef5f0f6cdf0266df31b1b", algorithm=MD5 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 13 [ 20]: Content-Length: 1102 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 14 [ 0]: [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 0 [ 3]: v=0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 1 [ 45]: o=- 3636434846 3636434846 IN IP4 172.17.1.166 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 2 [ 9]: s=cpc_med [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 3 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 4 [ 8]: b=AS:554 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 5 [ 5]: t=0 0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 6 [ 50]: m=audio 4000 RTP/SAVP 111 110 109 103 9 0 8 18 101 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 7 [ 10]: a=sendrecv [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 8 [ 25]: a=rtpmap:111 OPUS/48000/2 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 9 [ 47]: a=fmtp:111 maxplaybackrate=32000;useinbandfec=1 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 10 [ 23]: a=rtpmap:110 SILK/24000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 11 [ 25]: a=fmtp:110 useinbandfec=1 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 12 [ 23]: a=rtpmap:109 SILK/16000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 13 [ 25]: a=fmtp:109 useinbandfec=1 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 14 [ 25]: a=rtpmap:103 AMR-WB/16000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 15 [ 24]: a=fmtp:103 octet-align=0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 16 [ 20]: a=rtpmap:9 G722/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 17 [ 20]: a=rtpmap:0 PCMU/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 18 [ 20]: a=rtpmap:8 PCMA/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 19 [ 21]: a=rtpmap:18 G729/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 20 [ 19]: a=fmtp:18 annexb=no [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 21 [ 33]: a=rtpmap:101 telephone-event/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 22 [ 15]: a=fmtp:101 0-16 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 23 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7VPWc8um/bLNW2iWFR6tD6hciOqnvOYIQa9uMKYr [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 24 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:hNK5niRo8k8/Xbscu55VYsz7WPfg46mAtpRqR3L2 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 25 [ 28]: m=video 4002 RTP/SAVP 97 102 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 26 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 27 [ 13]: b=TIAS:512000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 28 [ 10]: a=sendrecv [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 29 [ 22]: a=rtpmap:97 H264/90000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 30 [ 33]: a=fmtp:97 profile-level-id=42800c [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 31 [ 22]: a=rtpmap:102 VP8/90000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 32 [ 17]: a=orient:portrait [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 33 [ 20]: a=rtcp-fb:* nack pli [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 34 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XmVadUHDw4fwentcubE+/bJ6DJis3xFXkLgBpFCs [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 35 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NDyHBo/dnJ8wUf4YDqOPZ/rWf+8zK/Ueyv7178Fk [2015-03-27 09:47:26] VERBOSE[15883] chan_sip.c: --- (14 headers 36 lines) --- [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Sending to 172.17.1.166:50199 (NAT) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Using INVITE request as basis request - 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found peer 'testClient' for 'testClient' from 172.17.1.166:50199 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6b4a054' [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Allocated port 13040 for RTP instance '0xb6b4a054' [2015-03-27 09:47:26] DEBUG[15883][C-00000000] pjsip: icess0xb6b41f7 ICE session created, comp_cnt=2, role is Unknown agent [2015-03-27 09:47:26] DEBUG[15883][C-00000000] pjsip: icess0xb6b41f7 Candidate 0 added: comp_id=1, type=host, foundation=Hac110301, addr=172.17.3.1:13040, base=172.17.3.1:13040, prio=0x7effffff (2130706431) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: RTP instance '0xb6b4a054' is setup and ready to go [2015-03-27 09:47:26] DEBUG[15883][C-00000000] pjsip: icess0xb6b41f7 Destroying ICE session 0xb6b41f74 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] pjsip: ice_session.c ICE session 0xb6b41f74 destroyed [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6b375bc' [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Allocated port 13008 for RTP instance '0xb6b375bc' [2015-03-27 09:47:26] DEBUG[15883][C-00000000] pjsip: icess0xb6b4234 ICE session created, comp_cnt=2, role is Unknown agent [2015-03-27 09:47:26] DEBUG[15883][C-00000000] pjsip: icess0xb6b4234 Candidate 0 added: comp_id=1, type=host, foundation=Hac110301, addr=172.17.3.1:13008, base=172.17.3.1:13008, prio=0x7effffff (2130706431) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: RTP instance '0xb6b375bc' is setup and ready to go [2015-03-27 09:47:26] DEBUG[15883][C-00000000] pjsip: icess0xb6b4234 Destroying ICE session 0xb6b4234c [2015-03-27 09:47:26] DEBUG[15883][C-00000000] pjsip: ice_session.c ICE session 0xb6b4234c destroyed [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6b375bc' [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] netsock2.c: Using SIP VIDEO CoS mark 6 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6b4a054' [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] netsock2.c: Using SIP RTP CoS mark 5 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Setting NAT on RTP to On [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Setting NAT on VRTP to On [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP o=- 3636434846 3636434846 IN IP4 172.17.1.166... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP s=cpc_med... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 172.17.1.166... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP b=AS:554... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP audio format 111 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 111 (0xb6b359ac) based on m type on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP audio format 110 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 110 (0xb6b359f4) based on m type on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP audio format 109 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 109 (0xb6b47ddc) based on m type on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP audio format 103 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 103 (0xb6b47e24) based on m type on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP audio format 9 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 9 (0xb6b47e6c) based on m type on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP audio format 0 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 0 (0xb6b47eb4) based on m type on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP audio format 8 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 8 (0xb6b47efc) based on m type on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP audio format 18 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 18 (0xb6b460f4) based on m type on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP audio format 101 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 101 (0xb6b4613c) based on m type on 0xb6a83ec0 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found audio description format OPUS for ID 111 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 OPUS/48000/2... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:111 maxplaybackrate=32000;useinbandfec=1... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Unsetting payload 110 on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found unknown media description format SILK for ID 110 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 SILK/24000... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:110 useinbandfec=1... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Unsetting payload 109 on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found unknown media description format SILK for ID 109 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:109 SILK/16000... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:109 useinbandfec=1... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Unsetting payload 103 on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found unknown media description format AMR-WB for ID 103 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:103 AMR-WB/16000... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:103 octet-align=0... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found audio description format G722 for ID 9 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found audio description format G729 for ID 18 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sdp_srtp.c: local_key64 qyvQ4xDAdE5JDL+tSNb/lV3ts2R7Xv/3vJOZ8kMd len 40 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_srtp.c: Adding new policy for SSRC 1944634453 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sdp_srtp.c: SRTP policy activated [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qyvQ4xDAdE5JDL+tSNb/lV3ts2R7Xv/3vJOZ8kMd [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7VPWc8um/bLNW2iWFR6tD6hciOqnvOYIQa9uMKYr... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:hNK5niRo8k8/Xbscu55VYsz7WPfg46mAtpRqR3L2... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP video format 97 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 97 (0xb6b3ac9c) based on m type on 0xb6a83ef8 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP video format 102 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 102 (0xb6b3ace4) based on m type on 0xb6a83ef8 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP c=IN IP4 172.17.1.166... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP b=TIAS:512000... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=sendrecv... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found video description format H264 for ID 97 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=rtpmap:97 H264/90000... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=fmtp:97 profile-level-id=42800c... OK. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found video description format VP8 for ID 102 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=rtpmap:102 VP8/90000... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=orient:portrait... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=rtcp-fb:* nack pli... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sdp_srtp.c: local_key64 wfvtyttJFPwOQ1r6ZEBW76wQnGTWqIofmuEOXXVL len 40 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_srtp.c: Adding new policy for SSRC 1932607279 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sdp_srtp.c: SRTP policy activated [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:wfvtyttJFPwOQ1r6ZEBW76wQnGTWqIofmuEOXXVL [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XmVadUHDw4fwentcubE+/bJ6DJis3xFXkLgBpFCs... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NDyHBo/dnJ8wUf4YDqOPZ/rWf+8zK/Ueyv7178Fk... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263|opus|h264), peer - audio=(ulaw|alaw|g722|g729|opus)/video=(h264|vp8)/text=(nothing), combined - (ulaw|alaw|opus|h264) [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b4a054' [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Peer audio RTP is at port 172.17.1.166:4000 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Copying payload 0 (0xb6b359f4) from 0xb6a83ec0 to 0xb6b4a200 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Copying payload 8 (0xb6b47ddc) from 0xb6a83ec0 to 0xb6b4a200 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Copying payload 9 (0xb6b359ac) from 0xb6a83ec0 to 0xb6b4a200 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Copying payload 18 (0xb6b461cc) from 0xb6a83ec0 to 0xb6b4a200 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Copying payload 101 (0xb6b460f4) from 0xb6a83ec0 to 0xb6b4a200 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Copying payload 111 (0xb6b46184) from 0xb6a83ec0 to 0xb6b4a200 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb6b4a054' [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b375bc' [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Peer video RTP is at port 172.17.1.166:4002 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Copying payload 97 (0xb6b3ad2c) from 0xb6a83ef8 to 0xb6b37768 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Copying payload 102 (0xb6b3ac9c) from 0xb6a83ef8 to 0xb6b37768 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: We're settling with these formats: (ulaw|alaw|opus|h264) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Checking SIP call limits for device testClient [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Updating call counter for incoming call [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Call from peer 'testClient' is 1 out of 5 [2015-03-27 09:47:26] DEBUG[15797] devicestate.c: No provider found, checking channel drivers for SIP - testClient [2015-03-27 09:47:26] DEBUG[15797] chan_sip.c: Checking device state for peer testClient [2015-03-27 09:47:26] DEBUG[15797] devicestate.c: Changing state for SIP/testClient - state 2 (In use) [2015-03-27 09:47:26] DEBUG[15799] app_queue.c: Extension '74721@hint' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2015-03-27 09:47:26] DEBUG[15905] app_queue.c: Device 'SIP/testClient' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Looking for 8501 in context-testClient (domain 172.17.3.1) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: *** Our native formats are (h264|ulaw) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: *** Joint capabilities are (ulaw|alaw|opus|h264) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: *** Our capabilities are (ulaw|alaw|gsm|h263|opus|h264) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: This channel can handle video! HOLLYWOOD next! [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] sip/route.c: sip_route_dump: route/path hop: [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: SIP/testClient-00000000: New call is still down.... Trying... [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: <--- Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPjlu5Evd-yK18umT6bsgRcpGSzxmHAXd.S;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2645 INVITE Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:26] DEBUG[15797] devicestate.c: No provider found, checking channel drivers for SIP - testClient [2015-03-27 09:47:26] DEBUG[15797] chan_sip.c: Checking device state for peer testClient [2015-03-27 09:47:26] DEBUG[15797] devicestate.c: Changing state for SIP/testClient - state 2 (In use) [2015-03-27 09:47:26] DEBUG[16232][C-00000000] pbx.c: Launching 'VoiceMail' [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] pbx.c: Executing [8501@context-testClient:1] VoiceMail("SIP/testClient-00000000", "1234,u") in new stack [2015-03-27 09:47:26] DEBUG[15797] devicestate.c: No provider found, checking channel drivers for SIP - testClient [2015-03-27 09:47:26] DEBUG[15797] chan_sip.c: Checking device state for peer testClient [2015-03-27 09:47:26] DEBUG[15797] devicestate.c: Changing state for SIP/testClient - state 2 (In use) [2015-03-27 09:47:26] DEBUG[16232][C-00000000] chan_sip.c: SIP answering channel: SIP/testClient-00000000 [2015-03-27 09:47:26] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-03-27 09:47:26] DEBUG[16232][C-00000000] chan_sip.c: Setting framing from config on incoming call [2015-03-27 09:47:26] DEBUG[16232][C-00000000] chan_sip.c: This call needs video offers! [2015-03-27 09:47:26] DEBUG[16232][C-00000000] chan_sip.c: ** Our capability: (ulaw|alaw|opus|h264) Video flag: False Text flag: True [2015-03-27 09:47:26] DEBUG[16232][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] chan_sip.c: Audio is at 13040 [2015-03-27 09:47:26] DEBUG[16232][C-00000000] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:wfvtyttJFPwOQ1r6ZEBW76wQnGTWqIofmuEOXXVL [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] chan_sip.c: Video is at 172.17.3.1:13008 [2015-03-27 09:47:26] DEBUG[16232][C-00000000] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qyvQ4xDAdE5JDL+tSNb/lV3ts2R7Xv/3vJOZ8kMd [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] chan_sip.c: Adding codec ulaw to SDP [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] chan_sip.c: Adding codec alaw to SDP [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] chan_sip.c: Adding codec opus to SDP [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] chan_sip.c: Adding video codec h264 to SDP [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] chan_sip.c: Adding codec gsm to SDP [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] chan_sip.c: Adding video codec h263 to SDP [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2015-03-27 09:47:26] DEBUG[16232][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [2015-03-27 09:47:26] DEBUG[16232][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw|opus|h264) [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] chan_sip.c: <--- Reliably Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPjlu5Evd-yK18umT6bsgRcpGSzxmHAXd.S;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: ;tag=as4b0ecfad Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2645 INVITE Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 783 v=0 o=root 1773947224 1773947224 IN IP4 172.17.3.1 s=Asterisk PBX SVN-branch-13-r433338M c=IN IP4 172.17.3.1 b=CT:384 t=0 0 m=audio 13040 RTP/SAVP 0 8 111 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 opus/48000/2 a=fmtp:111 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:60 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qyvQ4xDAdE5JDL+tSNb/lV3ts2R7Xv/3vJOZ8kMd m=video 13008 RTP/SAVP 97 34 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42800C a=rtpmap:34 H263/90000 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:wfvtyttJFPwOQ1r6ZEBW76wQnGTWqIofmuEOXXVL <------------> [2015-03-27 09:47:26] DEBUG[16232][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:26] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:26] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:26] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:26] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:26] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:26] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:26] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:26] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:26] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:26] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:26] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:26] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:26] DEBUG[16232][C-00000000] res_rtp_asterisk.c: 0xb6b5a180 -- Probation learning mode pass with source address 172.17.1.166:4000 [2015-03-27 09:47:26] VERBOSE[16232][C-00000000] res_rtp_asterisk.c: 0xb6b5a180 -- Probation passed - setting RTP source address to 172.17.1.166:4000 [2015-03-27 09:47:26] DEBUG[16232][C-00000000] app_voicemail.c: Before find_user [2015-03-27 09:47:26] DEBUG[16232][C-00000000] app_voicemail.c: /var/spool/asterisk/voicemail/default/1234/unavail doesn't exist, doing what we can [2015-03-27 09:47:26] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> ACK sip:8501@172.17.3.1:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjNU6hArw3V5lO5hHTfJgFtLKkNCKQWy9u;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: ;tag=as4b0ecfad Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2645 ACK Authorization: Digest username="testClient", realm="asterisk", nonce="6f5b9e6c", uri="sip:8501@172.17.3.1:5061;transport=TLS", response="284a42fb28a10ca7f7ad5841e2805b2f", algorithm=MD5 Content-Length: 0 <-------------> [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 0 [ 50]: ACK sip:8501@172.17.3.1:5061;transport=TLS SIP/2.0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjNU6hArw3V5lO5hHTfJgFtLKkNCKQWy9u;alias [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 4 [ 51]: To: ;tag=as4b0ecfad [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 5 [ 41]: Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 6 [ 14]: CSeq: 2645 ACK [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 7 [183]: Authorization: Digest username="testClient", realm="asterisk", nonce="6f5b9e6c", uri="sip:8501@172.17.3.1:5061;transport=TLS", response="284a42fb28a10ca7f7ad5841e2805b2f", algorithm=MD5 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [2015-03-27 09:47:26] VERBOSE[15883] chan_sip.c: --- (9 headers 0 lines) --- [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Stopping retransmission on '7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q' of Response 2645: Match Not Found [2015-03-27 09:47:26] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> INVITE sip:8501@172.17.3.1:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjRkQqtzik2gH0DeeJ1PllGjvjyxUFg6nl;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: ;tag=as4b0ecfad Contact: "SIP Test Client" Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2646 INVITE Authorization: Digest username="testClient", realm="asterisk", nonce="6f5b9e6c", uri="sip:8501@172.17.3.1", response="ba654b95ac5ef5f0f6cdf0266df31b1b", algorithm=MD5 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Type: application/sdp Content-Length: 730 v=0 o=- 3636434846 3636434847 IN IP4 172.17.1.166 s=cpc_med c=IN IP4 172.17.1.166 b=AS:554 t=0 0 m=audio 4000 RTP/SAVP 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7VPWc8um/bLNW2iWFR6tD6hciOqnvOYIQa9uMKYr a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:hNK5niRo8k8/Xbscu55VYsz7WPfg46mAtpRqR3L2 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv m=video 4002 RTP/SAVP 97 c=IN IP4 172.17.1.166 b=TIAS:512000 a=orient:portrait a=rtcp-fb:* nack pli a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XmVadUHDw4fwentcubE+/bJ6DJis3xFXkLgBpFCs a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NDyHBo/dnJ8wUf4YDqOPZ/rWf+8zK/Ueyv7178Fk a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42800c a=sendrecv <-------------> [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 0 [ 53]: INVITE sip:8501@172.17.3.1:5061;transport=TLS SIP/2.0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjRkQqtzik2gH0DeeJ1PllGjvjyxUFg6nl;alias [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 4 [ 51]: To: ;tag=as4b0ecfad [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 5 [ 70]: Contact: "SIP Test Client" [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 6 [ 41]: Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 7 [ 17]: CSeq: 2646 INVITE [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 8 [175]: Authorization: Digest username="testClient", realm="asterisk", nonce="6f5b9e6c", uri="sip:8501@172.17.3.1", response="ba654b95ac5ef5f0f6cdf0266df31b1b", algorithm=MD5 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 9 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 10 [ 39]: Supported: replaces, 100rel, norefersub [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 12 [ 19]: Content-Length: 730 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Header 13 [ 0]: [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 0 [ 3]: v=0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 1 [ 45]: o=- 3636434846 3636434847 IN IP4 172.17.1.166 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 2 [ 9]: s=cpc_med [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 3 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 4 [ 8]: b=AS:554 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 5 [ 5]: t=0 0 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 6 [ 27]: m=audio 4000 RTP/SAVP 0 101 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 7 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7VPWc8um/bLNW2iWFR6tD6hciOqnvOYIQa9uMKYr [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 8 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:hNK5niRo8k8/Xbscu55VYsz7WPfg46mAtpRqR3L2 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-16 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 12 [ 10]: a=sendrecv [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 13 [ 24]: m=video 4002 RTP/SAVP 97 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 14 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 15 [ 13]: b=TIAS:512000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 16 [ 17]: a=orient:portrait [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 17 [ 20]: a=rtcp-fb:* nack pli [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 18 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XmVadUHDw4fwentcubE+/bJ6DJis3xFXkLgBpFCs [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 19 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NDyHBo/dnJ8wUf4YDqOPZ/rWf+8zK/Ueyv7178Fk [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 20 [ 22]: a=rtpmap:97 H264/90000 [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 21 [ 33]: a=fmtp:97 profile-level-id=42800c [2015-03-27 09:47:26] DEBUG[15883] chan_sip.c: Body 22 [ 10]: a=sendrecv [2015-03-27 09:47:26] VERBOSE[15883] chan_sip.c: --- (13 headers 23 lines) --- [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Sending to 172.17.1.166:50199 (NAT) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP o=- 3636434846 3636434847 IN IP4 172.17.1.166... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP s=cpc_med... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 172.17.1.166... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP b=AS:554... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP audio format 0 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 0 (0xb6b507ec) based on m type on 0xb6a83ec0 [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP audio format 101 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 101 (0xb6b41b24) based on m type on 0xb6a83ec0 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sdp_srtp.c: SRTP remote key unchanged; maintaining current policy [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7VPWc8um/bLNW2iWFR6tD6hciOqnvOYIQa9uMKYr... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:hNK5niRo8k8/Xbscu55VYsz7WPfg46mAtpRqR3L2... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found RTP video format 97 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Setting payload 97 (0xb6b41b24) based on m type on 0xb6a83ef8 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP c=IN IP4 172.17.1.166... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP b=TIAS:512000... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=orient:portrait... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=rtcp-fb:* nack pli... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sdp_srtp.c: SRTP remote key unchanged; maintaining current policy [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XmVadUHDw4fwentcubE+/bJ6DJis3xFXkLgBpFCs... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NDyHBo/dnJ8wUf4YDqOPZ/rWf+8zK/Ueyv7178Fk... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Found video description format H264 for ID 97 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=rtpmap:97 H264/90000... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=fmtp:97 profile-level-id=42800c... OK. [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Processing media-level (video) SDP a=sendrecv... UNSUPPORTED OR FAILED. [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263|opus|h264), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264) [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b4a054' [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Peer audio RTP is at port 172.17.1.166:4000 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Copying payload 0 (0xb6b41b6c) from 0xb6a83ec0 to 0xb6b4a200 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Copying payload 101 (0xb6b507ec) from 0xb6a83ec0 to 0xb6b4a200 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb6b4a054' [2015-03-27 09:47:26] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b375bc' [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Peer video RTP is at port 172.17.1.166:4002 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] rtp_engine.c: Copying payload 97 (0xb6b7b634) from 0xb6a83ef8 to 0xb6b37768 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: We're settling with these formats: (ulaw|h264) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: We have an owner, now see if we need to change this call [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw|h264), old nativeformats (h264|ulaw) [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Got a SIP re-invite for call 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: SIP/testClient-00000000: This call is UP.... [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: <--- Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPjRkQqtzik2gH0DeeJ1PllGjvjyxUFg6nl;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: ;tag=as4b0ecfad Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2646 INVITE Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Setting framing from config on incoming call [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: This call needs video offers! [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: ** Our capability: (ulaw|h264) Video flag: False Text flag: True [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Audio is at 13040 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:wfvtyttJFPwOQ1r6ZEBW76wQnGTWqIofmuEOXXVL [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Video is at 172.17.3.1:13008 [2015-03-27 09:47:26] DEBUG[15883][C-00000000] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qyvQ4xDAdE5JDL+tSNb/lV3ts2R7Xv/3vJOZ8kMd [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Adding codec ulaw to SDP [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Adding video codec h264 to SDP [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Adding codec alaw to SDP [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Adding codec gsm to SDP [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Adding video codec h263 to SDP [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Adding codec opus to SDP [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|h264) [2015-03-27 09:47:26] VERBOSE[15883][C-00000000] chan_sip.c: <--- Reliably Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPjRkQqtzik2gH0DeeJ1PllGjvjyxUFg6nl;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: ;tag=as4b0ecfad Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2646 INVITE Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 632 v=0 o=root 1773947224 1773947225 IN IP4 172.17.3.1 s=Asterisk PBX SVN-branch-13-r433338M c=IN IP4 172.17.3.1 b=CT:384 t=0 0 m=audio 13040 RTP/SAVP 0 8 3 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:60 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qyvQ4xDAdE5JDL+tSNb/lV3ts2R7Xv/3vJOZ8kMd m=video 13008 RTP/SAVP 97 34 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42800C a=rtpmap:34 H263/90000 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:wfvtyttJFPwOQ1r6ZEBW76wQnGTWqIofmuEOXXVL <------------> [2015-03-27 09:47:26] DEBUG[15883][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:26] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:27] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> ACK sip:8501@172.17.3.1:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjNBZynEhrCQ5-cn.2kV1MO0f.raoMJviv;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: ;tag=as4b0ecfad Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2646 ACK Authorization: Digest username="testClient", realm="asterisk", nonce="6f5b9e6c", uri="sip:8501@172.17.3.1:5061;transport=TLS", response="284a42fb28a10ca7f7ad5841e2805b2f", algorithm=MD5 Content-Length: 0 <-------------> [2015-03-27 09:47:27] DEBUG[15883] chan_sip.c: Header 0 [ 50]: ACK sip:8501@172.17.3.1:5061;transport=TLS SIP/2.0 [2015-03-27 09:47:27] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjNBZynEhrCQ5-cn.2kV1MO0f.raoMJviv;alias [2015-03-27 09:47:27] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:27] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q [2015-03-27 09:47:27] DEBUG[15883] chan_sip.c: Header 4 [ 51]: To: ;tag=as4b0ecfad [2015-03-27 09:47:27] DEBUG[15883] chan_sip.c: Header 5 [ 41]: Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:27] DEBUG[15883] chan_sip.c: Header 6 [ 14]: CSeq: 2646 ACK [2015-03-27 09:47:27] DEBUG[15883] chan_sip.c: Header 7 [183]: Authorization: Digest username="testClient", realm="asterisk", nonce="6f5b9e6c", uri="sip:8501@172.17.3.1:5061;transport=TLS", response="284a42fb28a10ca7f7ad5841e2805b2f", algorithm=MD5 [2015-03-27 09:47:27] DEBUG[15883] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [2015-03-27 09:47:27] VERBOSE[15883] chan_sip.c: --- (9 headers 0 lines) --- [2015-03-27 09:47:27] DEBUG[15883][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2015-03-27 09:47:27] DEBUG[15883][C-00000000] chan_sip.c: Stopping retransmission on '7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q' of Response 2646: Match Not Found [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Ooh, format changed from none to ulaw [2015-03-27 09:47:27] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:47:27] VERBOSE[16232][C-00000000] file.c: Playing 'vm-theperson.gsm' (language 'en') [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: 0xb6b5cad0 -- Probation learning mode pass with source address 172.17.1.166:4002 [2015-03-27 09:47:27] VERBOSE[16232][C-00000000] res_rtp_asterisk.c: 0xb6b5cad0 -- Probation passed - setting RTP source address to 172.17.1.166:4002 [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: 0xb6b5a180 -- Probation learning mode pass with source address 172.17.1.166:4000 [2015-03-27 09:47:27] VERBOSE[16232][C-00000000] res_rtp_asterisk.c: 0xb6b5a180 -- Probation passed - setting RTP source address to 172.17.1.166:4000 [2015-03-27 09:47:27] DEBUG[15866] chan_iax2.c: ip callno count decremented to 7 for 92.226.232.223 [2015-03-27 09:47:27] DEBUG[15866] chan_iax2.c: ip callno count decremented to 6 for 92.226.232.223 [2015-03-27 09:47:27] DEBUG[15866] chan_iax2.c: ip callno count decremented to 3 for 77.37.11.180 [2015-03-27 09:47:27] DEBUG[15868] chan_iax2.c: schedule decrement of callno used for 92.226.232.223 in 60 seconds [2015-03-27 09:47:27] DEBUG[15870] chan_iax2.c: schedule decrement of callno used for 92.226.232.223 in 60 seconds [2015-03-27 09:47:27] DEBUG[15872] chan_iax2.c: schedule decrement of callno used for 77.37.11.180 in 60 seconds [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:27] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:27] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:28] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:28] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:28] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:28] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:28] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:28] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second ..... [2015-03-27 09:47:31] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:47:31] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:31] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:31] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:31] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:31] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:31] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:47:32] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:32] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:32] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second ... [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:32] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second ... [2015-03-27 09:47:33] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:47:33] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:33] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:33] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:33] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:33] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:33] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:33] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:33] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:33] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second ... [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second ... [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:34] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second ... [2015-03-27 09:47:35] DEBUG[15866] chan_iax2.c: ip callno count decremented to 2 for 85.25.208.239 [2015-03-27 09:47:35] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:35] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:35] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:35] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:35] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:35] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second ... [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second ... [2015-03-27 09:47:36] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:47:36] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:36] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:36] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:36] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:47:36] VERBOSE[16232][C-00000000] file.c: Playing 'vm-isunavail.gsm' (language 'en') [2015-03-27 09:47:36] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:36] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:36] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:47:37] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:37] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:37] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:47:37] DEBUG[15876] chan_iax2.c: ip callno count incremented to 7 for 92.226.232.223 [2015-03-27 09:47:37] DEBUG[15877] chan_iax2.c: ip callno count incremented to 8 for 92.226.232.223 [2015-03-27 09:47:37] DEBUG[15868] chan_iax2.c: ip callno count incremented to 4 for 77.37.11.180 [2015-03-27 09:47:38] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:38] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:38] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:38] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:38] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:38] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:47:38] VERBOSE[16232][C-00000000] file.c: Playing 'vm-intro.gsm' (language 'en') [2015-03-27 09:47:38] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:47:38] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:38] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:38] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:41] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:47:41] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:41] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:41] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:41] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:41] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:41] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:47:42] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:42] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:42] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:47:43] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:47:43] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:43] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:43] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:43] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:43] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:43] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:43] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:43] DEBUG[16232][C-00000000] app.c: Locked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:43] DEBUG[16232][C-00000000] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:43] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format gsm [2015-03-27 09:47:43] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:47:43] VERBOSE[16232][C-00000000] file.c: Playing 'beep.gsm' (language 'en') [2015-03-27 09:47:43] DEBUG[15866] chan_iax2.c: ip callno count decremented to 1 for 85.25.208.239 [2015-03-27 09:47:43] DEBUG[15872] chan_iax2.c: ip callno count incremented to 2 for 85.25.208.239 [2015-03-27 09:47:43] DEBUG[15873] chan_iax2.c: schedule decrement of callno used for 85.25.208.239 in 60 seconds [2015-03-27 09:47:43] DEBUG[15873] chan_iax2.c: Peer 12346178: got pong, lastms 27, historicms 27, maxms 2000 [2015-03-27 09:47:44] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:44] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:44] DEBUG[16232][C-00000000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:44] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to write format ulaw [2015-03-27 09:47:44] VERBOSE[16232][C-00000000] app_voicemail.c: Recording the message [2015-03-27 09:47:44] DEBUG[16232][C-00000000] app.c: play_and_record: , /var/spool/asterisk/voicemail/default/1234/tmp/XWULlK, 'wav49|gsm|wav|ulaw' [2015-03-27 09:47:44] DEBUG[16232][C-00000000] app.c: Recording Formats: sfmts=wav49 [2015-03-27 09:47:44] VERBOSE[16232][C-00000000] app.c: x=0, open writing: /var/spool/asterisk/voicemail/default/1234/tmp/XWULlK format: wav49, 0x9d3ed54 [2015-03-27 09:47:44] VERBOSE[16232][C-00000000] app.c: x=1, open writing: /var/spool/asterisk/voicemail/default/1234/tmp/XWULlK format: gsm, 0x9ae6d04 [2015-03-27 09:47:44] VERBOSE[16232][C-00000000] app.c: x=2, open writing: /var/spool/asterisk/voicemail/default/1234/tmp/XWULlK format: wav, 0xa3f0ab4 [2015-03-27 09:47:44] VERBOSE[16232][C-00000000] app.c: x=3, open writing: /var/spool/asterisk/voicemail/default/1234/tmp/XWULlK format: ulaw, 0x9ae1e04 [2015-03-27 09:47:44] DEBUG[16232][C-00000000] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [2015-03-27 09:47:44] DEBUG[16232][C-00000000] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [2015-03-27 09:47:44] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to read format slin [2015-03-27 09:47:44] DEBUG[16232][C-00000000] chan_sip.c: Strict routing enforced for session 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:44] VERBOSE[16232][C-00000000] chan_sip.c: Reliably Transmitting (NAT) to 172.17.1.166:50199: INFO sip:testClient@172.17.1.166:50199;transport=TLS;ob SIP/2.0 Via: SIP/2.0/TLS 172.17.3.1:5061;branch=z9hG4bK68898a7b;rport Max-Forwards: 70 From: ;tag=as4b0ecfad To: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q Contact: Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 102 INFO User-Agent: Asterisk PBX SVN-branch-13-r433338M Content-Type: application/media_control+xml Content-Length: 205 --- [2015-03-27 09:47:44] DEBUG[16232][C-00000000] chan_sip.c: Trying to put 'INFO sip:ma' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:44] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.17.3.1:5061;rport=5061;received=172.17.3.1;branch=z9hG4bK68898a7b Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q From: ;tag=as4b0ecfad To: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q CSeq: 102 INFO Content-Length: 0 <-------------> [2015-03-27 09:47:44] DEBUG[15883] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2015-03-27 09:47:44] DEBUG[15883] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/TLS 172.17.3.1:5061;rport=5061;received=172.17.3.1;branch=z9hG4bK68898a7b [2015-03-27 09:47:44] DEBUG[15883] chan_sip.c: Header 2 [ 41]: Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:44] DEBUG[15883] chan_sip.c: Header 3 [ 53]: From: ;tag=as4b0ecfad [2015-03-27 09:47:44] DEBUG[15883] chan_sip.c: Header 4 [ 88]: To: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q [2015-03-27 09:47:44] DEBUG[15883] chan_sip.c: Header 5 [ 14]: CSeq: 102 INFO [2015-03-27 09:47:44] DEBUG[15883] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [2015-03-27 09:47:44] VERBOSE[15883] chan_sip.c: --- (7 headers 0 lines) --- [2015-03-27 09:47:44] DEBUG[16232][C-00000000] file.c: Opened video output file [2015-03-27 09:47:45] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:47:45] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:45] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:45] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:46] DEBUG[16231] threadpool.c: Worker thread idle timeout reached. Dying. [2015-03-27 09:47:46] DEBUG[15785] threadpool.c: Destroying worker thread 14 [2015-03-27 09:47:46] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:46] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:46] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:47:47] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:47] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:47] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:47:47] DEBUG[15866] chan_iax2.c: ip callno count decremented to 7 for 92.226.232.223 [2015-03-27 09:47:47] DEBUG[15866] chan_iax2.c: ip callno count decremented to 6 for 92.226.232.223 [2015-03-27 09:47:47] DEBUG[15866] chan_iax2.c: ip callno count decremented to 3 for 77.37.11.180 [2015-03-27 09:47:47] DEBUG[15876] chan_iax2.c: schedule decrement of callno used for 92.226.232.223 in 60 seconds [2015-03-27 09:47:47] DEBUG[15868] chan_iax2.c: schedule decrement of callno used for 92.226.232.223 in 60 seconds [2015-03-27 09:47:47] DEBUG[15870] chan_iax2.c: schedule decrement of callno used for 77.37.11.180 in 60 seconds [2015-03-27 09:47:48] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:47:48] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:48] DEBUG[16232][C-00000000] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:48] DEBUG[16232][C-00000000] acl.c: Attached to given IP address [2015-03-27 09:47:49] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> BYE sip:8501@172.17.3.1:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjXQvWm7QCWco6za.glpGwT6zqF5hNjXuy;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: ;tag=as4b0ecfad Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2647 BYE Authorization: Digest username="testClient", realm="asterisk", nonce="6f5b9e6c", uri="sip:8501@172.17.3.1:5061;transport=TLS", response="4271631d5286708f4ed406c27828d80a", algorithm=MD5 User-Agent: Bria iOS 3.3.0 Content-Length: 0 <-------------> [2015-03-27 09:47:49] DEBUG[15883] chan_sip.c: Header 0 [ 50]: BYE sip:8501@172.17.3.1:5061;transport=TLS SIP/2.0 [2015-03-27 09:47:49] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjXQvWm7QCWco6za.glpGwT6zqF5hNjXuy;alias [2015-03-27 09:47:49] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:49] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q [2015-03-27 09:47:49] DEBUG[15883] chan_sip.c: Header 4 [ 51]: To: ;tag=as4b0ecfad [2015-03-27 09:47:49] DEBUG[15883] chan_sip.c: Header 5 [ 41]: Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:49] DEBUG[15883] chan_sip.c: Header 6 [ 14]: CSeq: 2647 BYE [2015-03-27 09:47:49] DEBUG[15883] chan_sip.c: Header 7 [183]: Authorization: Digest username="testClient", realm="asterisk", nonce="6f5b9e6c", uri="sip:8501@172.17.3.1:5061;transport=TLS", response="4271631d5286708f4ed406c27828d80a", algorithm=MD5 [2015-03-27 09:47:49] DEBUG[15883] chan_sip.c: Header 8 [ 26]: User-Agent: Bria iOS 3.3.0 [2015-03-27 09:47:49] DEBUG[15883] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [2015-03-27 09:47:49] VERBOSE[15883] chan_sip.c: --- (10 headers 0 lines) --- [2015-03-27 09:47:49] DEBUG[15883][C-00000000] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [2015-03-27 09:47:49] DEBUG[15883][C-00000000] chan_sip.c: Initializing initreq for method BYE - callid 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:49] VERBOSE[15883][C-00000000] chan_sip.c: Sending to 172.17.1.166:50199 (NAT) [2015-03-27 09:47:49] DEBUG[15883][C-00000000] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:49] DEBUG[15785] threadpool.c: Increasing threadpool stasis-core's size by 1 [2015-03-27 09:47:49] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b4a054' [2015-03-27 09:47:49] DEBUG[15883][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b375bc' [2015-03-27 09:47:49] VERBOSE[16232][C-00000000] app.c: User hung up [2015-03-27 09:47:49] DEBUG[16232][C-00000000] channel.c: Set channel SIP/testClient-00000000 to read format ulaw [2015-03-27 09:47:49] VERBOSE[15883][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q' in 32000 ms (Method: BYE) [2015-03-27 09:47:49] DEBUG[15883][C-00000000] chan_sip.c: Received bye, issuing owner hangup [2015-03-27 09:47:49] VERBOSE[15883][C-00000000] chan_sip.c: <--- Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPjXQvWm7QCWco6za.glpGwT6zqF5hNjXuy;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=ECUu5r60zJ7UTevm3l0rIHOChLeBdE.Q To: ;tag=as4b0ecfad Call-ID: 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q CSeq: 2647 BYE Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [2015-03-27 09:47:49] DEBUG[15883][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:49] DEBUG[16232][C-00000000] app.c: Locked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:49] DEBUG[16232][C-00000000] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:49] DEBUG[16232][C-00000000] app_voicemail.c: Attaching file '/var/spool/asterisk/voicemail/default/1234/INBOX/msg0000', format 'WAV', uservm is '0', global is 2048 [2015-03-27 09:47:49] DEBUG[16232][C-00000000] app_voicemail.c: Sent mail to xxxxxx with command '/usr/sbin/sendmail -t' [2015-03-27 09:47:49] DEBUG[16232][C-00000000] pbx.c: Spawn extension (context-testClient,8501,1) exited non-zero on 'SIP/testClient-00000000' [2015-03-27 09:47:49] VERBOSE[16232][C-00000000] pbx.c: Spawn extension (context-testClient, 8501, 1) exited non-zero on 'SIP/testClient-00000000' [2015-03-27 09:47:49] DEBUG[16232][C-00000000] channel.c: Soft-Hanging (0x10) up channel 'SIP/testClient-00000000' [2015-03-27 09:47:49] DEBUG[16232][C-00000000] channel.c: Hanging up channel 'SIP/testClient-00000000' [2015-03-27 09:47:49] DEBUG[16232][C-00000000] chan_sip.c: Hangup call SIP/testClient-00000000, SIP callid 7-JYxzEdO6THthk2iHHdrbMqTSNbOv7Q [2015-03-27 09:47:49] DEBUG[16232][C-00000000] chan_sip.c: update_call_counter(testClient) - decrement call limit counter on hangup [2015-03-27 09:47:49] DEBUG[16232][C-00000000] chan_sip.c: Updating call counter for incoming call [2015-03-27 09:47:49] DEBUG[16232][C-00000000] chan_sip.c: Call from peer 'testClient' removed from call limit 5 [2015-03-27 09:47:49] DEBUG[15797] devicestate.c: No provider found, checking channel drivers for SIP - testClient [2015-03-27 09:47:49] DEBUG[15797] chan_sip.c: Checking device state for peer testClient [2015-03-27 09:47:49] DEBUG[15797] devicestate.c: Changing state for SIP/testClient - state 1 (Not in use) [2015-03-27 09:47:49] DEBUG[15799] app_queue.c: Extension '74721@hint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2015-03-27 09:47:49] DEBUG[15905] app_queue.c: Device 'SIP/testClient' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [2015-03-27 09:47:49] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b4a054' [2015-03-27 09:47:49] DEBUG[16232][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b375bc' [2015-03-27 09:47:49] DEBUG[15798] cdr.c: Finalized CDR for SIP/testClient-00000000 - start 1427446046.938007 answer 1427446046.954980 end 1427446069.840257 dispo ANSWERED [2015-03-27 09:47:49] DEBUG[15798] cdr_pgsql.c: Inserting a CDR record: [INSERT INTO cdr ("calldate","clid","src","dst","dcontext","channel","lastapp","lastdata","duration","billsec","disposition","amaflags","uniqueid") VALUES ('2015-03-27 09:47:26','"SIP Test Client" ','testClient','8501','context-testClient','SIP/testClient-00000000','VoiceMail','1234,u',22,22,'ANSWERED',3,'1427446046.1')] [2015-03-27 09:47:49] DEBUG[15797] devicestate.c: No provider found, checking channel drivers for SIP - testClient [2015-03-27 09:47:49] DEBUG[15797] chan_sip.c: Checking device state for peer testClient [2015-03-27 09:47:49] DEBUG[15797] devicestate.c: Changing state for SIP/testClient - state 1 (Not in use) [2015-03-27 09:47:52] DEBUG[15861] chan_sip.c: Auto destroying SIP dialog '19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB' [2015-03-27 09:47:52] DEBUG[15861] chan_sip.c: Destroying SIP dialog 19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB [2015-03-27 09:47:52] VERBOSE[15861] chan_sip.c: Really destroying SIP dialog '19vmoo50mZPgsxdA3cMXXCZPGz5jvbaB' Method: REGISTER [2015-03-27 09:47:55] DEBUG[15891] pbx_dundi.c: Register us as '00:40:63:e3:96:05' to '00:e0:4d:6c:93:72' [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> INVITE sip:8502@172.17.3.1 SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjo5j5P14nxdjMruLNf4VX6sENd93AQl87;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 To: Contact: "SIP Test Client" Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp CSeq: 270 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria iOS 3.3.0 Content-Type: application/sdp Content-Length: 1102 v=0 o=- 3636434876 3636434876 IN IP4 172.17.1.166 s=cpc_med c=IN IP4 172.17.1.166 b=AS:554 t=0 0 m=audio 4004 RTP/SAVP 111 110 109 103 9 0 8 18 101 a=sendrecv a=rtpmap:111 OPUS/48000/2 a=fmtp:111 maxplaybackrate=32000;useinbandfec=1 a=rtpmap:110 SILK/24000 a=fmtp:110 useinbandfec=1 a=rtpmap:109 SILK/16000 a=fmtp:109 useinbandfec=1 a=rtpmap:103 AMR-WB/16000 a=fmtp:103 octet-align=0 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R/TEw6JCKsB9oRPImfyEtUUNUMCOoShzQJpSkgkY a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OEZa0kP5MV1EwChYuTfAUXcHOJYvqAG+6DTZe3XW m=video 4006 RTP/SAVP 97 102 c=IN IP4 172.17.1.166 b=TIAS:512000 a=sendrecv a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42800c a=rtpmap:102 VP8/90000 a=orient:portrait a=rtcp-fb:* nack pli a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pt/WEtFWco+ri6rYfW8pYfrK13xcYJ9JUhE1N2Dw a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CLkfTOZZznVeEOcnWXetseAlJflNRAS5JjTJrpAy <-------------> [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 0 [ 45]: INVITE sip:8502@172.17.3.1 SIP/2.0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjo5j5P14nxdjMruLNf4VX6sENd93AQl87;alias [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 4 [ 36]: To: [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 5 [ 70]: Contact: "SIP Test Client" [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 6 [ 41]: Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 7 [ 16]: CSeq: 270 INVITE [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 8 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 9 [ 39]: Supported: replaces, 100rel, norefersub [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 10 [ 26]: User-Agent: Bria iOS 3.3.0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 12 [ 20]: Content-Length: 1102 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 13 [ 0]: [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 0 [ 3]: v=0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 1 [ 45]: o=- 3636434876 3636434876 IN IP4 172.17.1.166 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 2 [ 9]: s=cpc_med [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 3 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 4 [ 8]: b=AS:554 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 5 [ 5]: t=0 0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 6 [ 50]: m=audio 4004 RTP/SAVP 111 110 109 103 9 0 8 18 101 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 7 [ 10]: a=sendrecv [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 8 [ 25]: a=rtpmap:111 OPUS/48000/2 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 9 [ 47]: a=fmtp:111 maxplaybackrate=32000;useinbandfec=1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 10 [ 23]: a=rtpmap:110 SILK/24000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 11 [ 25]: a=fmtp:110 useinbandfec=1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 12 [ 23]: a=rtpmap:109 SILK/16000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 13 [ 25]: a=fmtp:109 useinbandfec=1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 14 [ 25]: a=rtpmap:103 AMR-WB/16000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 15 [ 24]: a=fmtp:103 octet-align=0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 16 [ 20]: a=rtpmap:9 G722/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 17 [ 20]: a=rtpmap:0 PCMU/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 18 [ 20]: a=rtpmap:8 PCMA/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 19 [ 21]: a=rtpmap:18 G729/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 20 [ 19]: a=fmtp:18 annexb=no [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 21 [ 33]: a=rtpmap:101 telephone-event/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 22 [ 15]: a=fmtp:101 0-16 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 23 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R/TEw6JCKsB9oRPImfyEtUUNUMCOoShzQJpSkgkY [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 24 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OEZa0kP5MV1EwChYuTfAUXcHOJYvqAG+6DTZe3XW [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 25 [ 28]: m=video 4006 RTP/SAVP 97 102 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 26 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 27 [ 13]: b=TIAS:512000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 28 [ 10]: a=sendrecv [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 29 [ 22]: a=rtpmap:97 H264/90000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 30 [ 33]: a=fmtp:97 profile-level-id=42800c [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 31 [ 22]: a=rtpmap:102 VP8/90000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 32 [ 17]: a=orient:portrait [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 33 [ 20]: a=rtcp-fb:* nack pli [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 34 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pt/WEtFWco+ri6rYfW8pYfrK13xcYJ9JUhE1N2Dw [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 35 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CLkfTOZZznVeEOcnWXetseAlJflNRAS5JjTJrpAy [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: --- (13 headers 36 lines) --- [2015-03-27 09:47:56] DEBUG[15883] acl.c: For destination '172.17.1.166', our source address is '172.17.3.1'. [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Setting AST_TRANSPORT_TLS with address 172.17.3.1:5061 [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: Sending to 172.17.1.166:50199 (no NAT) [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Allocating new SIP dialog for jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp - INVITE (No RTP) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, norefersub" [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sip/reqresp_parser.c: Found SIP option: -replaces- [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sip/reqresp_parser.c: Matched SIP option: replaces [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sip/reqresp_parser.c: Found SIP option: -100rel- [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sip/reqresp_parser.c: Matched SIP option: 100rel [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sip/reqresp_parser.c: Found SIP option: -norefersub- [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sip/reqresp_parser.c: Matched SIP option: norefersub [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Sending to 172.17.1.166:50199 (no NAT) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Using INVITE request as basis request - jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found peer 'testClient' for 'testClient' from 172.17.1.166:50199 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: <--- Reliably Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPjo5j5P14nxdjMruLNf4VX6sENd93AQl87;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 To: ;tag=as36ce5077 Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp CSeq: 270 INVITE Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a538560" Content-Length: 0 <------------> [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 401' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog 'jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp' in 32000 ms (Method: INVITE) [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> ACK sip:8502@172.17.3.1 SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjo5j5P14nxdjMruLNf4VX6sENd93AQl87;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 To: ;tag=as36ce5077 Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp CSeq: 270 ACK Content-Length: 0 <-------------> [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 0 [ 42]: ACK sip:8502@172.17.3.1 SIP/2.0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjo5j5P14nxdjMruLNf4VX6sENd93AQl87;alias [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 4 [ 51]: To: ;tag=as36ce5077 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 5 [ 41]: Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 6 [ 13]: CSeq: 270 ACK [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: --- (8 headers 0 lines) --- [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Stopping retransmission on 'jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp' of Response 270: Match Not Found [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> INVITE sip:8502@172.17.3.1 SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjaQWMQdp.UQkujYCpTP.iCTUwawOVQCt.;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 To: Contact: "SIP Test Client" Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp CSeq: 271 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria iOS 3.3.0 Authorization: Digest username="testClient", realm="asterisk", nonce="3a538560", uri="sip:8502@172.17.3.1", response="3d1f18351b30868a47bd805c043d3285", algorithm=MD5 Content-Type: application/sdp Content-Length: 1102 v=0 o=- 3636434876 3636434876 IN IP4 172.17.1.166 s=cpc_med c=IN IP4 172.17.1.166 b=AS:554 t=0 0 m=audio 4004 RTP/SAVP 111 110 109 103 9 0 8 18 101 a=sendrecv a=rtpmap:111 OPUS/48000/2 a=fmtp:111 maxplaybackrate=32000;useinbandfec=1 a=rtpmap:110 SILK/24000 a=fmtp:110 useinbandfec=1 a=rtpmap:109 SILK/16000 a=fmtp:109 useinbandfec=1 a=rtpmap:103 AMR-WB/16000 a=fmtp:103 octet-align=0 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R/TEw6JCKsB9oRPImfyEtUUNUMCOoShzQJpSkgkY a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OEZa0kP5MV1EwChYuTfAUXcHOJYvqAG+6DTZe3XW m=video 4006 RTP/SAVP 97 102 c=IN IP4 172.17.1.166 b=TIAS:512000 a=sendrecv a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42800c a=rtpmap:102 VP8/90000 a=orient:portrait a=rtcp-fb:* nack pli a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pt/WEtFWco+ri6rYfW8pYfrK13xcYJ9JUhE1N2Dw a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CLkfTOZZznVeEOcnWXetseAlJflNRAS5JjTJrpAy <-------------> [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 0 [ 45]: INVITE sip:8502@172.17.3.1 SIP/2.0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjaQWMQdp.UQkujYCpTP.iCTUwawOVQCt.;alias [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 4 [ 36]: To: [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 5 [ 70]: Contact: "SIP Test Client" [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 6 [ 41]: Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 7 [ 16]: CSeq: 271 INVITE [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 8 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 9 [ 39]: Supported: replaces, 100rel, norefersub [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 10 [ 26]: User-Agent: Bria iOS 3.3.0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 11 [175]: Authorization: Digest username="testClient", realm="asterisk", nonce="3a538560", uri="sip:8502@172.17.3.1", response="3d1f18351b30868a47bd805c043d3285", algorithm=MD5 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 13 [ 20]: Content-Length: 1102 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 14 [ 0]: [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 0 [ 3]: v=0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 1 [ 45]: o=- 3636434876 3636434876 IN IP4 172.17.1.166 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 2 [ 9]: s=cpc_med [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 3 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 4 [ 8]: b=AS:554 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 5 [ 5]: t=0 0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 6 [ 50]: m=audio 4004 RTP/SAVP 111 110 109 103 9 0 8 18 101 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 7 [ 10]: a=sendrecv [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 8 [ 25]: a=rtpmap:111 OPUS/48000/2 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 9 [ 47]: a=fmtp:111 maxplaybackrate=32000;useinbandfec=1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 10 [ 23]: a=rtpmap:110 SILK/24000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 11 [ 25]: a=fmtp:110 useinbandfec=1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 12 [ 23]: a=rtpmap:109 SILK/16000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 13 [ 25]: a=fmtp:109 useinbandfec=1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 14 [ 25]: a=rtpmap:103 AMR-WB/16000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 15 [ 24]: a=fmtp:103 octet-align=0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 16 [ 20]: a=rtpmap:9 G722/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 17 [ 20]: a=rtpmap:0 PCMU/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 18 [ 20]: a=rtpmap:8 PCMA/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 19 [ 21]: a=rtpmap:18 G729/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 20 [ 19]: a=fmtp:18 annexb=no [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 21 [ 33]: a=rtpmap:101 telephone-event/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 22 [ 15]: a=fmtp:101 0-16 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 23 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R/TEw6JCKsB9oRPImfyEtUUNUMCOoShzQJpSkgkY [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 24 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OEZa0kP5MV1EwChYuTfAUXcHOJYvqAG+6DTZe3XW [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 25 [ 28]: m=video 4006 RTP/SAVP 97 102 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 26 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 27 [ 13]: b=TIAS:512000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 28 [ 10]: a=sendrecv [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 29 [ 22]: a=rtpmap:97 H264/90000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 30 [ 33]: a=fmtp:97 profile-level-id=42800c [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 31 [ 22]: a=rtpmap:102 VP8/90000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 32 [ 17]: a=orient:portrait [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 33 [ 20]: a=rtcp-fb:* nack pli [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 34 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pt/WEtFWco+ri6rYfW8pYfrK13xcYJ9JUhE1N2Dw [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 35 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CLkfTOZZznVeEOcnWXetseAlJflNRAS5JjTJrpAy [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: --- (14 headers 36 lines) --- [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Sending to 172.17.1.166:50199 (NAT) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Using INVITE request as basis request - jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found peer 'testClient' for 'testClient' from 172.17.1.166:50199 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6b344a4' [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_rtp_asterisk.c: Allocated port 13026 for RTP instance '0xb6b344a4' [2015-03-27 09:47:56] DEBUG[15883][C-00000001] pjsip: icess0xb6b5316 ICE session created, comp_cnt=2, role is Unknown agent [2015-03-27 09:47:56] DEBUG[15883][C-00000001] pjsip: icess0xb6b5316 Candidate 0 added: comp_id=1, type=host, foundation=Hac110301, addr=172.17.3.1:13026, base=172.17.3.1:13026, prio=0x7effffff (2130706431) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: RTP instance '0xb6b344a4' is setup and ready to go [2015-03-27 09:47:56] DEBUG[15883][C-00000001] pjsip: icess0xb6b5316 Destroying ICE session 0xb6b5316c [2015-03-27 09:47:56] DEBUG[15883][C-00000001] pjsip: ice_session.c ICE session 0xb6b5316c destroyed [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6b3d69c' [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_rtp_asterisk.c: Allocated port 13028 for RTP instance '0xb6b3d69c' [2015-03-27 09:47:56] DEBUG[15883][C-00000001] pjsip: icess0xb6b55ab ICE session created, comp_cnt=2, role is Unknown agent [2015-03-27 09:47:56] DEBUG[15883][C-00000001] pjsip: icess0xb6b55ab Candidate 0 added: comp_id=1, type=host, foundation=Hac110301, addr=172.17.3.1:13028, base=172.17.3.1:13028, prio=0x7effffff (2130706431) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: RTP instance '0xb6b3d69c' is setup and ready to go [2015-03-27 09:47:56] DEBUG[15883][C-00000001] pjsip: icess0xb6b55ab Destroying ICE session 0xb6b55abc [2015-03-27 09:47:56] DEBUG[15883][C-00000001] pjsip: ice_session.c ICE session 0xb6b55abc destroyed [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6b3d69c' [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] netsock2.c: Using SIP VIDEO CoS mark 6 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6b344a4' [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] netsock2.c: Using SIP RTP CoS mark 5 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Setting NAT on RTP to On [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Setting NAT on VRTP to On [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP o=- 3636434876 3636434876 IN IP4 172.17.1.166... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP s=cpc_med... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 172.17.1.166... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP b=AS:554... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP audio format 111 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 111 (0xb6b7a08c) based on m type on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP audio format 110 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 110 (0xb6b7a0d4) based on m type on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP audio format 109 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 109 (0xb6b75d34) based on m type on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP audio format 103 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 103 (0xb6b75d7c) based on m type on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP audio format 9 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 9 (0xb6b41ab4) based on m type on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP audio format 0 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 0 (0xb6b41afc) based on m type on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP audio format 8 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 8 (0xb6b7527c) based on m type on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP audio format 18 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 18 (0xb6b752c4) based on m type on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP audio format 101 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 101 (0xb6b50734) based on m type on 0xb6a83ec0 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found audio description format OPUS for ID 111 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 OPUS/48000/2... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:111 maxplaybackrate=32000;useinbandfec=1... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Unsetting payload 110 on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found unknown media description format SILK for ID 110 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 SILK/24000... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:110 useinbandfec=1... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Unsetting payload 109 on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found unknown media description format SILK for ID 109 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:109 SILK/16000... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:109 useinbandfec=1... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Unsetting payload 103 on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found unknown media description format AMR-WB for ID 103 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:103 AMR-WB/16000... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:103 octet-align=0... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found audio description format G722 for ID 9 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found audio description format G729 for ID 18 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sdp_srtp.c: local_key64 IZzbPuU47KnaPMzFL1GvAvn0a/wQZ2Sz+s3yu1Bf len 40 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_srtp.c: Adding new policy for SSRC 832203682 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sdp_srtp.c: SRTP policy activated [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:IZzbPuU47KnaPMzFL1GvAvn0a/wQZ2Sz+s3yu1Bf [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R/TEw6JCKsB9oRPImfyEtUUNUMCOoShzQJpSkgkY... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OEZa0kP5MV1EwChYuTfAUXcHOJYvqAG+6DTZe3XW... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP video format 97 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 97 (0xb6b50734) based on m type on 0xb6a83ef8 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP video format 102 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 102 (0xb6b7b58c) based on m type on 0xb6a83ef8 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP c=IN IP4 172.17.1.166... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP b=TIAS:512000... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=sendrecv... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found video description format H264 for ID 97 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=rtpmap:97 H264/90000... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=fmtp:97 profile-level-id=42800c... OK. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found video description format VP8 for ID 102 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=rtpmap:102 VP8/90000... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=orient:portrait... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=rtcp-fb:* nack pli... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sdp_srtp.c: local_key64 lPoCg8ps50oFJ0ZiZ55/mTQsab+Q5Kztu33dYfqz len 40 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_srtp.c: Adding new policy for SSRC 1461137126 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sdp_srtp.c: SRTP policy activated [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lPoCg8ps50oFJ0ZiZ55/mTQsab+Q5Kztu33dYfqz [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pt/WEtFWco+ri6rYfW8pYfrK13xcYJ9JUhE1N2Dw... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CLkfTOZZznVeEOcnWXetseAlJflNRAS5JjTJrpAy... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263|opus|h264), peer - audio=(ulaw|alaw|g722|g729|opus)/video=(h264|vp8)/text=(nothing), combined - (ulaw|alaw|opus|h264) [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b344a4' [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Peer audio RTP is at port 172.17.1.166:4004 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Copying payload 0 (0xb6b41ab4) from 0xb6a83ec0 to 0xb6b34650 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Copying payload 8 (0xb6b7a0d4) from 0xb6a83ec0 to 0xb6b34650 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Copying payload 9 (0xb6b7a08c) from 0xb6a83ec0 to 0xb6b34650 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Copying payload 18 (0xb6b7527c) from 0xb6a83ec0 to 0xb6b34650 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Copying payload 101 (0xb6b41afc) from 0xb6a83ec0 to 0xb6b34650 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Copying payload 111 (0xb6b5077c) from 0xb6a83ec0 to 0xb6b34650 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb6b344a4' [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b3d69c' [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Peer video RTP is at port 172.17.1.166:4006 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Copying payload 97 (0xb6b7b5d4) from 0xb6a83ef8 to 0xb6b3d848 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Copying payload 102 (0xb6b50734) from 0xb6a83ef8 to 0xb6b3d848 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: We're settling with these formats: (ulaw|alaw|opus|h264) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Checking SIP call limits for device testClient [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Updating call counter for incoming call [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Call from peer 'testClient' is 1 out of 5 [2015-03-27 09:47:56] DEBUG[15797] devicestate.c: No provider found, checking channel drivers for SIP - testClient [2015-03-27 09:47:56] DEBUG[15797] chan_sip.c: Checking device state for peer testClient [2015-03-27 09:47:56] DEBUG[15797] devicestate.c: Changing state for SIP/testClient - state 2 (In use) [2015-03-27 09:47:56] DEBUG[15799] app_queue.c: Extension '74721@hint' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2015-03-27 09:47:56] DEBUG[15905] app_queue.c: Device 'SIP/testClient' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Looking for 8502 in context-testClient (domain 172.17.3.1) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: *** Our native formats are (h264|ulaw) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: *** Joint capabilities are (ulaw|alaw|opus|h264) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: *** Our capabilities are (ulaw|alaw|gsm|h263|opus|h264) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: This channel can handle video! HOLLYWOOD next! [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] sip/route.c: sip_route_dump: route/path hop: [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: SIP/testClient-00000001: New call is still down.... Trying... [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: <--- Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPjaQWMQdp.UQkujYCpTP.iCTUwawOVQCt.;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 To: Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp CSeq: 271 INVITE Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 100' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:56] DEBUG[15797] devicestate.c: No provider found, checking channel drivers for SIP - testClient [2015-03-27 09:47:56] DEBUG[15797] chan_sip.c: Checking device state for peer testClient [2015-03-27 09:47:56] DEBUG[15797] devicestate.c: Changing state for SIP/testClient - state 2 (In use) [2015-03-27 09:47:56] DEBUG[16240][C-00000001] pbx.c: Launching 'VoiceMailMain' [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] pbx.c: Executing [8502@context-testClient:1] VoiceMailMain("SIP/testClient-00000001", "1234,su") in new stack [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app_voicemail.c: Before ast_answer [2015-03-27 09:47:56] DEBUG[15797] devicestate.c: No provider found, checking channel drivers for SIP - testClient [2015-03-27 09:47:56] DEBUG[15797] chan_sip.c: Checking device state for peer testClient [2015-03-27 09:47:56] DEBUG[15797] devicestate.c: Changing state for SIP/testClient - state 2 (In use) [2015-03-27 09:47:56] DEBUG[16240][C-00000001] chan_sip.c: SIP answering channel: SIP/testClient-00000001 [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Setting the marker bit due to a source update [2015-03-27 09:47:56] DEBUG[16240][C-00000001] chan_sip.c: Setting framing from config on incoming call [2015-03-27 09:47:56] DEBUG[16240][C-00000001] chan_sip.c: This call needs video offers! [2015-03-27 09:47:56] DEBUG[16240][C-00000001] chan_sip.c: ** Our capability: (ulaw|alaw|opus|h264) Video flag: False Text flag: True [2015-03-27 09:47:56] DEBUG[16240][C-00000001] chan_sip.c: ** Our prefcodec: (nothing) [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] chan_sip.c: Audio is at 13026 [2015-03-27 09:47:56] DEBUG[16240][C-00000001] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lPoCg8ps50oFJ0ZiZ55/mTQsab+Q5Kztu33dYfqz [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] chan_sip.c: Video is at 172.17.3.1:13028 [2015-03-27 09:47:56] DEBUG[16240][C-00000001] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:IZzbPuU47KnaPMzFL1GvAvn0a/wQZ2Sz+s3yu1Bf [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] chan_sip.c: Adding codec ulaw to SDP [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] chan_sip.c: Adding codec alaw to SDP [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] chan_sip.c: Adding codec opus to SDP [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] chan_sip.c: Adding video codec h264 to SDP [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] chan_sip.c: Adding codec gsm to SDP [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] chan_sip.c: Adding video codec h263 to SDP [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2015-03-27 09:47:56] DEBUG[16240][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [2015-03-27 09:47:56] DEBUG[16240][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw|opus|h264) [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] chan_sip.c: <--- Reliably Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPjaQWMQdp.UQkujYCpTP.iCTUwawOVQCt.;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 To: ;tag=as6cd9d1f3 Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp CSeq: 271 INVITE Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 783 v=0 o=root 1806988343 1806988343 IN IP4 172.17.3.1 s=Asterisk PBX SVN-branch-13-r433338M c=IN IP4 172.17.3.1 b=CT:384 t=0 0 m=audio 13026 RTP/SAVP 0 8 111 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 opus/48000/2 a=fmtp:111 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:60 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:IZzbPuU47KnaPMzFL1GvAvn0a/wQZ2Sz+s3yu1Bf m=video 13028 RTP/SAVP 97 34 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42800C a=rtpmap:34 H263/90000 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lPoCg8ps50oFJ0ZiZ55/mTQsab+Q5Kztu33dYfqz <------------> [2015-03-27 09:47:56] DEBUG[16240][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 200' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: 0xb6b50808 -- Probation learning mode pass with source address 172.17.1.166:4004 [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] res_rtp_asterisk.c: 0xb6b50808 -- Probation passed - setting RTP source address to 172.17.1.166:4004 [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app_voicemail.c: After vm_authenticate [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app_voicemail.c: Before open_mailbox [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Locked path '/var/spool/asterisk/voicemail/default/1234/Old' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/1234/Old' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Locked path '/var/spool/asterisk/voicemail/default/1234/Old' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/1234/Old' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app_voicemail.c: Number of old messages: 0 [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Locked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Locked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app_voicemail.c: /var/spool/asterisk/voicemail/default/1234/INBOX map[0] = 1, count = 1 [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app_voicemail.c: Number of new messages: 1 [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Locked path '/var/spool/asterisk/voicemail/default/1234/Urgent' [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> ACK sip:8502@172.17.3.1:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjWi7ZnYsYWgr1TEFnx5dMXoD-HPcYCOgr;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 To: ;tag=as6cd9d1f3 Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp CSeq: 271 ACK Authorization: Digest username="testClient", realm="asterisk", nonce="3a538560", uri="sip:8502@172.17.3.1:5061;transport=TLS", response="e16f150911cac860e0f87aff3725ca5d", algorithm=MD5 Content-Length: 0 <-------------> [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 0 [ 50]: ACK sip:8502@172.17.3.1:5061;transport=TLS SIP/2.0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjWi7ZnYsYWgr1TEFnx5dMXoD-HPcYCOgr;alias [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 4 [ 51]: To: ;tag=as6cd9d1f3 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 5 [ 41]: Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 6 [ 13]: CSeq: 271 ACK [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 7 [183]: Authorization: Digest username="testClient", realm="asterisk", nonce="3a538560", uri="sip:8502@172.17.3.1:5061;transport=TLS", response="e16f150911cac860e0f87aff3725ca5d", algorithm=MD5 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: --- (9 headers 0 lines) --- [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Stopping retransmission on 'jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp' of Response 271: Match Not Found [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/1234/Urgent' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Locked path '/var/spool/asterisk/voicemail/default/1234/Urgent' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/1234/Urgent' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app_voicemail.c: Number of urgent messages: 0 [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> INVITE sip:8502@172.17.3.1:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPj6JCbBH.-kC0saztSCRUb.h4JJKOFy-N9;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 To: ;tag=as6cd9d1f3 Contact: "SIP Test Client" Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp CSeq: 272 INVITE Authorization: Digest username="testClient", realm="asterisk", nonce="3a538560", uri="sip:8502@172.17.3.1", response="3d1f18351b30868a47bd805c043d3285", algorithm=MD5 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub Content-Type: application/sdp Content-Length: 730 v=0 o=- 3636434876 3636434877 IN IP4 172.17.1.166 s=cpc_med c=IN IP4 172.17.1.166 b=AS:554 t=0 0 m=audio 4004 RTP/SAVP 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R/TEw6JCKsB9oRPImfyEtUUNUMCOoShzQJpSkgkY a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OEZa0kP5MV1EwChYuTfAUXcHOJYvqAG+6DTZe3XW a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv m=video 4006 RTP/SAVP 97 c=IN IP4 172.17.1.166 b=TIAS:512000 a=orient:portrait a=rtcp-fb:* nack pli a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pt/WEtFWco+ri6rYfW8pYfrK13xcYJ9JUhE1N2Dw a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CLkfTOZZznVeEOcnWXetseAlJflNRAS5JjTJrpAy a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42800c a=sendrecv <-------------> [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 0 [ 53]: INVITE sip:8502@172.17.3.1:5061;transport=TLS SIP/2.0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPj6JCbBH.-kC0saztSCRUb.h4JJKOFy-N9;alias [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 4 [ 51]: To: ;tag=as6cd9d1f3 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 5 [ 70]: Contact: "SIP Test Client" [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 6 [ 41]: Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 7 [ 16]: CSeq: 272 INVITE [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 8 [175]: Authorization: Digest username="testClient", realm="asterisk", nonce="3a538560", uri="sip:8502@172.17.3.1", response="3d1f18351b30868a47bd805c043d3285", algorithm=MD5 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 9 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 10 [ 39]: Supported: replaces, 100rel, norefersub [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 12 [ 19]: Content-Length: 730 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 13 [ 0]: [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 0 [ 3]: v=0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 1 [ 45]: o=- 3636434876 3636434877 IN IP4 172.17.1.166 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 2 [ 9]: s=cpc_med [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 3 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 4 [ 8]: b=AS:554 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 5 [ 5]: t=0 0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 6 [ 27]: m=audio 4004 RTP/SAVP 0 101 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 7 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R/TEw6JCKsB9oRPImfyEtUUNUMCOoShzQJpSkgkY [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 8 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OEZa0kP5MV1EwChYuTfAUXcHOJYvqAG+6DTZe3XW [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Locked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Locked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app_voicemail.c: /var/spool/asterisk/voicemail/default/1234/INBOX map[0] = 1, count = 1 [2015-03-27 09:47:56] DEBUG[16240][C-00000001] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/1234/INBOX' [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-16 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 12 [ 10]: a=sendrecv [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 13 [ 24]: m=video 4006 RTP/SAVP 97 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 14 [ 21]: c=IN IP4 172.17.1.166 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 15 [ 13]: b=TIAS:512000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 16 [ 17]: a=orient:portrait [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 17 [ 20]: a=rtcp-fb:* nack pli [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 18 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pt/WEtFWco+ri6rYfW8pYfrK13xcYJ9JUhE1N2Dw [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 19 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CLkfTOZZznVeEOcnWXetseAlJflNRAS5JjTJrpAy [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 20 [ 22]: a=rtpmap:97 H264/90000 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 21 [ 33]: a=fmtp:97 profile-level-id=42800c [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Body 22 [ 10]: a=sendrecv [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: --- (13 headers 23 lines) --- [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Sending to 172.17.1.166:50199 (NAT) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Initializing initreq for method INVITE - callid jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP o=- 3636434876 3636434877 IN IP4 172.17.1.166... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP s=cpc_med... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 172.17.1.166... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP b=AS:554... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP audio format 0 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 0 (0xb6b4037c) based on m type on 0xb6a83ec0 [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP audio format 101 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 101 (0xb6b7ecc4) based on m type on 0xb6a83ec0 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sdp_srtp.c: SRTP remote key unchanged; maintaining current policy [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:R/TEw6JCKsB9oRPImfyEtUUNUMCOoShzQJpSkgkY... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OEZa0kP5MV1EwChYuTfAUXcHOJYvqAG+6DTZe3XW... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found RTP video format 97 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Setting payload 97 (0xb6b7ecc4) based on m type on 0xb6a83ef8 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP c=IN IP4 172.17.1.166... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP b=TIAS:512000... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=orient:portrait... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=rtcp-fb:* nack pli... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sdp_srtp.c: SRTP remote key unchanged; maintaining current policy [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pt/WEtFWco+ri6rYfW8pYfrK13xcYJ9JUhE1N2Dw... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CLkfTOZZznVeEOcnWXetseAlJflNRAS5JjTJrpAy... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Found video description format H264 for ID 97 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=rtpmap:97 H264/90000... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=fmtp:97 profile-level-id=42800c... OK. [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Processing media-level (video) SDP a=sendrecv... UNSUPPORTED OR FAILED. [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263|opus|h264), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264) [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b344a4' [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Peer audio RTP is at port 172.17.1.166:4004 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Copying payload 0 (0xb6b7ed0c) from 0xb6a83ec0 to 0xb6b34650 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Copying payload 101 (0xb6b4037c) from 0xb6a83ec0 to 0xb6b34650 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb6b344a4' [2015-03-27 09:47:56] DEBUG[15883][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb6b3d69c' [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Peer video RTP is at port 172.17.1.166:4006 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] rtp_engine.c: Copying payload 97 (0xb6b7ed54) from 0xb6a83ef8 to 0xb6b3d848 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: We're settling with these formats: (ulaw|h264) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: We have an owner, now see if we need to change this call [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw|h264), old nativeformats (h264|ulaw) [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Got a SIP re-invite for call jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: SIP/testClient-00000001: This call is UP.... [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: <--- Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPj6JCbBH.-kC0saztSCRUb.h4JJKOFy-N9;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 To: ;tag=as6cd9d1f3 Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp CSeq: 272 INVITE Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 100' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Setting framing from config on incoming call [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: This call needs video offers! [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: ** Our capability: (ulaw|h264) Video flag: False Text flag: True [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: ** Our prefcodec: (nothing) [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Audio is at 13026 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lPoCg8ps50oFJ0ZiZ55/mTQsab+Q5Kztu33dYfqz [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Video is at 172.17.3.1:13028 [2015-03-27 09:47:56] DEBUG[15883][C-00000001] sdp_srtp.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:IZzbPuU47KnaPMzFL1GvAvn0a/wQZ2Sz+s3yu1Bf [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Adding codec ulaw to SDP [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Adding video codec h264 to SDP [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Adding codec alaw to SDP [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Adding codec gsm to SDP [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Adding video codec h263 to SDP [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Adding codec opus to SDP [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|h264) [2015-03-27 09:47:56] VERBOSE[15883][C-00000001] chan_sip.c: <--- Reliably Transmitting (NAT) to 172.17.1.166:50199 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 172.17.1.166:50199;branch=z9hG4bKPj6JCbBH.-kC0saztSCRUb.h4JJKOFy-N9;alias;received=172.17.1.166;rport=50199 From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 To: ;tag=as6cd9d1f3 Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp CSeq: 272 INVITE Server: Asterisk PBX SVN-branch-13-r433338M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 632 v=0 o=root 1806988343 1806988344 IN IP4 172.17.3.1 s=Asterisk PBX SVN-branch-13-r433338M c=IN IP4 172.17.3.1 b=CT:384 t=0 0 m=audio 13026 RTP/SAVP 0 8 3 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:60 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:IZzbPuU47KnaPMzFL1GvAvn0a/wQZ2Sz+s3yu1Bf m=video 13028 RTP/SAVP 97 34 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42800C a=rtpmap:34 H263/90000 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lPoCg8ps50oFJ0ZiZ55/mTQsab+Q5Kztu33dYfqz <------------> [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 200' onto TLS socket destined for 172.17.1.166:50199 [2015-03-27 09:47:56] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: <--- SIP read from TLS:172.17.1.166:50199 ---> ACK sip:8502@172.17.3.1:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjCHFgktYOTPvkGKXl-BRnlTTUWdvgOES1;alias Max-Forwards: 70 From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 To: ;tag=as6cd9d1f3 Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp CSeq: 272 ACK Authorization: Digest username="testClient", realm="asterisk", nonce="3a538560", uri="sip:8502@172.17.3.1:5061;transport=TLS", response="e16f150911cac860e0f87aff3725ca5d", algorithm=MD5 Content-Length: 0 <-------------> [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 0 [ 50]: ACK sip:8502@172.17.3.1:5061;transport=TLS SIP/2.0 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/TLS 172.17.1.166:50199;rport;branch=z9hG4bKPjCHFgktYOTPvkGKXl-BRnlTTUWdvgOES1;alias [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 3 [ 90]: From: "SIP Test Client" ;tag=I7N0ouERqJNYZbxNWrp71tJXvrFC9Yr1 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 4 [ 51]: To: ;tag=as6cd9d1f3 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 5 [ 41]: Call-ID: jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 6 [ 13]: CSeq: 272 ACK [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 7 [183]: Authorization: Digest username="testClient", realm="asterisk", nonce="3a538560", uri="sip:8502@172.17.3.1:5061;transport=TLS", response="e16f150911cac860e0f87aff3725ca5d", algorithm=MD5 [2015-03-27 09:47:56] DEBUG[15883] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [2015-03-27 09:47:56] VERBOSE[15883] chan_sip.c: --- (9 headers 0 lines) --- [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [2015-03-27 09:47:56] DEBUG[15883][C-00000001] chan_sip.c: Stopping retransmission on 'jl.L7Zt5BWOmEeJRQ8sbSS-08lju0YIp' of Response 272: Match Not Found [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Ooh, format changed from none to ulaw [2015-03-27 09:47:56] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] file.c: Playing 'vm-youhave.gsm' (language 'en') [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: 0xb6b53158 -- Probation learning mode pass with source address 172.17.1.166:4006 [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] res_rtp_asterisk.c: 0xb6b53158 -- Probation passed - setting RTP source address to 172.17.1.166:4006 [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:56] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:56] DEBUG[16240][C-00000001] res_rtp_asterisk.c: 0xb6b50808 -- Probation learning mode pass with source address 172.17.1.166:4004 [2015-03-27 09:47:56] VERBOSE[16240][C-00000001] res_rtp_asterisk.c: 0xb6b50808 -- Probation passed - setting RTP source address to 172.17.1.166:4004 [2015-03-27 09:47:57] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:57] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:57] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:57] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:57] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [2015-03-27 09:47:57] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:57] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:57] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:57] DEBUG[15874] chan_iax2.c: ip callno count incremented to 7 for 92.226.232.223 [2015-03-27 09:47:57] DEBUG[15875] chan_iax2.c: ip callno count incremented to 8 for 92.226.232.223 [2015-03-27 09:47:57] DEBUG[15876] chan_iax2.c: ip callno count incremented to 4 for 77.37.11.180 [2015-03-27 09:47:57] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:57] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:57] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:57] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:47:57] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:47:57] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:47:57] VERBOSE[16240][C-00000001] file.c: Playing 'digits/1.gsm' (language 'en') [2015-03-27 09:47:57] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:47:57] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:57] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:47:57] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:47:58] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:58] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:58] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:58] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:47:58] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:47:58] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:47:58] VERBOSE[16240][C-00000001] file.c: Playing 'vm-INBOX.gsm' (language 'en') [2015-03-27 09:47:58] NOTICE[15861] chan_sip.c: -- Re-registration for 1888433@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2015-03-27 09:47:58] DEBUG[15861] srv.c: ast_get_srv: SRV lookup for '_sip._udp.sipgate.de' mapped to host sipgate.de, port 5060 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Allocating new SIP dialog for 280e17b757468575633fc0c959093c2f@sipgate.de - REGISTER (No RTP) [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2015-03-27 09:47:58] DEBUG[15861] acl.c: For destination '217.10.79.9', our source address is '172.17.3.1'. [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Target address 217.10.79.9:5060 is not local, substituting externaddr [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Setting AST_TRANSPORT_UDP with address 217.84.193.49:5060 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Scheduled a registration timeout for sipgate.de id #135 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: >>> Re-using Auth data for 1888433@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Initializing initreq for method REGISTER - callid 280e17b757468575633fc0c959093c2f@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 0 [ 31]: REGISTER sip:sipgate.de SIP/2.0 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 217.84.193.49:5060;branch=z9hG4bK6c0de048;rport [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 3 [ 45]: From: ;tag=as75eb6389 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 4 [ 28]: To: [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 5 [ 52]: Call-ID: 280e17b757468575633fc0c959093c2f@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 6 [ 18]: CSeq: 111 REGISTER [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 7 [ 26]: Supported: replaces, timer [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-13-r433338M [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 9 [192]: Authorization: Digest username="1888433", realm="sipgate.de", algorithm=MD5, uri="sip:sipgate.de", nonce="55151b315a820fdf7c07879d66a4f74da51ab7fd", response="5032d2dbbc45379a2c33b664508e3646" [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 10 [ 12]: Expires: 120 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 11 [ 41]: Contact: [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: REGISTER attempt 1 to 1888433@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #136 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 217.10.79.9:5060 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 217.84.193.49:5060;branch=z9hG4bK6c0de048;rport=61532 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 2 [ 45]: From: ;tag=as75eb6389 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 3 [ 70]: To: ;tag=12353b526ced3f57e916dd04ee4917bc.7e1a [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 4 [ 52]: Call-ID: 280e17b757468575633fc0c959093c2f@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 5 [ 18]: CSeq: 111 REGISTER [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 6 [134]: Contact: ;expires=112, ;expires=120;received="sip:217.84.193.49:61532" [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #136 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Stopping retransmission on '280e17b757468575633fc0c959093c2f@sipgate.de' of Request 111: Match Found [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Registration successful [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Cancelling timeout 135 [2015-03-27 09:47:58] NOTICE[15861] chan_sip.c: Outbound Registration: Expiry for sipgate.de is 120 sec (Scheduling reregistration in 105 s) [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Destroying SIP dialog 280e17b757468575633fc0c959093c2f@sipgate.de [2015-03-27 09:47:58] NOTICE[15861] chan_sip.c: -- Re-registration for 1276363@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2015-03-27 09:47:58] DEBUG[15861] srv.c: ast_get_srv: SRV lookup for '_sip._udp.sipgate.de' mapped to host sipgate.de, port 5060 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Allocating new SIP dialog for 2220e40225ba2e3f06d3777969f1fa66@sipgate.de - REGISTER (No RTP) [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2015-03-27 09:47:58] DEBUG[15861] acl.c: For destination '217.10.79.9', our source address is '172.17.3.1'. [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Target address 217.10.79.9:5060 is not local, substituting externaddr [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Setting AST_TRANSPORT_UDP with address 217.84.193.49:5060 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Scheduled a registration timeout for sipgate.de id #138 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: >>> Re-using Auth data for 1276363@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Initializing initreq for method REGISTER - callid 2220e40225ba2e3f06d3777969f1fa66@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 0 [ 31]: REGISTER sip:sipgate.de SIP/2.0 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 217.84.193.49:5060;branch=z9hG4bK27fcf9bf;rport [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 3 [ 45]: From: ;tag=as35079217 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 4 [ 28]: To: [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 5 [ 52]: Call-ID: 2220e40225ba2e3f06d3777969f1fa66@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 6 [ 18]: CSeq: 111 REGISTER [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 7 [ 26]: Supported: replaces, timer [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-13-r433338M [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 9 [192]: Authorization: Digest username="1276363", realm="sipgate.de", algorithm=MD5, uri="sip:sipgate.de", nonce="55151b317a9985336c23266215a94fa7b3236bae", response="95b58f84aac736561662aa71ea62f7a1" [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 10 [ 12]: Expires: 120 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 11 [ 41]: Contact: [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: REGISTER attempt 1 to 1276363@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #139 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 217.10.79.9:5060 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 217.84.193.49:5060;branch=z9hG4bK27fcf9bf;rport=61532 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 2 [ 45]: From: ;tag=as35079217 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 3 [ 70]: To: ;tag=fbf1d80521ea9f98078b6998e7669f9b.3a66 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 4 [ 52]: Call-ID: 2220e40225ba2e3f06d3777969f1fa66@sipgate.de [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 5 [ 18]: CSeq: 111 REGISTER [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 6 [133]: Contact: ;expires=86, ;expires=120;received="sip:217.84.193.49:61532" [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #139 [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Stopping retransmission on '2220e40225ba2e3f06d3777969f1fa66@sipgate.de' of Request 111: Match Found [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Registration successful [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Cancelling timeout 138 [2015-03-27 09:47:58] NOTICE[15861] chan_sip.c: Outbound Registration: Expiry for sipgate.de is 120 sec (Scheduling reregistration in 105 s) [2015-03-27 09:47:58] DEBUG[15861] chan_sip.c: Destroying SIP dialog 2220e40225ba2e3f06d3777969f1fa66@sipgate.de [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:47:59] VERBOSE[16240][C-00000001] file.c: Playing 'vm-message.gsm' (language 'en') [2015-03-27 09:47:59] NOTICE[15861] chan_sip.c: -- Re-registration for 41325102981@free3.voipgateway.org [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Allocating new SIP dialog for 23d9e34f4c77845c04f6a44e6e3ea44e@free3.voipgateway.org - REGISTER (No RTP) [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: OBPROXY: Not applying OBproxy to this call [2015-03-27 09:47:59] DEBUG[15861] acl.c: For destination '212.117.203.32', our source address is '172.17.3.1'. [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Target address 212.117.203.32:5060 is not local, substituting externaddr [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Setting AST_TRANSPORT_UDP with address 217.84.193.49:5060 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Scheduled a registration timeout for free3.voipgateway.org id #141 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: >>> Re-using Auth data for 41325102981@free3.voipgateway.org [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Initializing initreq for method REGISTER - callid 23d9e34f4c77845c04f6a44e6e3ea44e@free3.voipgateway.org [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 0 [ 42]: REGISTER sip:free3.voipgateway.org SIP/2.0 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 217.84.193.49:5060;branch=z9hG4bK3caa263e;rport [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 3 [ 60]: From: ;tag=as1e2b93df [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 4 [ 43]: To: [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 5 [ 63]: Call-ID: 23d9e34f4c77845c04f6a44e6e3ea44e@free3.voipgateway.org [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 6 [ 18]: CSeq: 112 REGISTER [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 7 [ 26]: Supported: replaces, timer [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 8 [ 47]: User-Agent: Asterisk PBX SVN-branch-13-r433338M [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 9 [221]: Authorization: Digest username="41325102981", realm="free3.voipgateway.org", algorithm=MD5, uri="sip:free3.voipgateway.org", nonce="1427445978:e3eff507d0055d4634c5da5b815303d3", response="ea9928f7ddd9f641866e534584f14afa" [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 10 [ 12]: Expires: 120 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 11 [ 45]: Contact: [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: REGISTER attempt 1 to 41325102981@free3.voipgateway.org [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #142 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Trying to put 'REGISTER si' onto UDP socket destined for 212.117.203.32:5060 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 217.84.193.49:5060;branch=z9hG4bK3caa263e;rport=61532 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 2 [ 72]: Contact: ;expires=3553 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 3 [ 72]: Contact: ;expires=3600 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 4 [ 56]: To: ;tag=bb275002 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 5 [ 60]: From: ;tag=as1e2b93df [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 6 [ 63]: Call-ID: 23d9e34f4c77845c04f6a44e6e3ea44e@free3.voipgateway.org [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 7 [ 18]: CSeq: 112 REGISTER [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 8 [ 35]: Date: Fri, 27 Mar 2015 08:48:03 GMT [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 9 [ 51]: PortaBilling: available-funds:24.03500 currency:CHF [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #142 [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Stopping retransmission on '23d9e34f4c77845c04f6a44e6e3ea44e@free3.voipgateway.org' of Request 112: Match Found [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Registration successful [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Cancelling timeout 141 [2015-03-27 09:47:59] NOTICE[15861] chan_sip.c: Outbound Registration: Expiry for free3.voipgateway.org is 120 sec (Scheduling reregistration in 105 s) [2015-03-27 09:47:59] DEBUG[15861] chan_sip.c: Destroying SIP dialog 23d9e34f4c77845c04f6a44e6e3ea44e@free3.voipgateway.org [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:47:59] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:47:59] VERBOSE[16240][C-00000001] file.c: Playing 'vm-onefor.gsm' (language 'en') [2015-03-27 09:48:00] NOTICE[15891] pbx_dundi.c: Max retries exceeded to host '172.17.1.101:4520' msg 0 on call 17971 [2015-03-27 09:48:00] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 49 (1), at 172.17.1.166:4004 [2015-03-27 09:48:00] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Creating END DTMF Frame: 49 (1), at 172.17.1.166:4004 [2015-03-27 09:48:00] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:00] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:00] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:48:00] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:48:00] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:48:00] VERBOSE[16240][C-00000001] file.c: Playing 'vm-first.gsm' (language 'en') [2015-03-27 09:48:01] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:01] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:01] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:01] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:48:01] DEBUG[16240][C-00000001] config.c: Parsing /var/spool/asterisk/voicemail/default/1234/INBOX/msg0000.txt [2015-03-27 09:48:01] VERBOSE[16240][C-00000001] config.c: Parsing '/var/spool/asterisk/voicemail/default/1234/INBOX/msg0000.txt': Found [2015-03-27 09:48:01] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:48:01] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:48:01] VERBOSE[16240][C-00000001] file.c: Playing 'vm-message.gsm' (language 'en') [2015-03-27 09:48:01] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:48:01] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:01] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:01] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:48:01] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:01] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:01] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:48:01] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:01] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:01] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:48:02] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:02] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:02] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:02] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:48:02] DEBUG[16240][C-00000001] say.c: Parsing ' (offset 0) in 'vm-received' q 'digits/at' IMp [2015-03-27 09:48:02] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:48:02] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:48:02] VERBOSE[16240][C-00000001] file.c: Playing 'vm-received.gsm' (language 'en') [2015-03-27 09:48:02] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:48:02] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:02] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:02] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:48:03] DEBUG[16240][C-00000001] say.c: Parsing (offset 13) in 'vm-received' q 'digits/at' IMp [2015-03-27 09:48:03] DEBUG[16240][C-00000001] say.c: Parsing q (offset 14) in 'vm-received' q 'digits/at' IMp [2015-03-27 09:48:03] DEBUG[16240][C-00000001] say.c: Parsing (offset 15) in 'vm-received' q 'digits/at' IMp [2015-03-27 09:48:03] DEBUG[16240][C-00000001] say.c: Parsing ' (offset 16) in 'vm-received' q 'digits/at' IMp [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:48:03] VERBOSE[16240][C-00000001] file.c: Playing 'digits/at.gsm' (language 'en') [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:48:03] DEBUG[16240][C-00000001] say.c: Parsing (offset 27) in 'vm-received' q 'digits/at' IMp [2015-03-27 09:48:03] DEBUG[16240][C-00000001] say.c: Parsing I (offset 28) in 'vm-received' q 'digits/at' IMp [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:48:03] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:48:03] VERBOSE[16240][C-00000001] file.c: Playing 'digits/9.gsm' (language 'en') [2015-03-27 09:48:04] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:04] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:04] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:04] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:48:04] DEBUG[16240][C-00000001] say.c: Parsing M (offset 29) in 'vm-received' q 'digits/at' IMp [2015-03-27 09:48:04] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:48:04] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:48:04] VERBOSE[16240][C-00000001] file.c: Playing 'digits/40.gsm' (language 'en') [2015-03-27 09:48:05] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:05] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:05] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:05] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:48:05] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:48:05] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:48:05] VERBOSE[16240][C-00000001] file.c: Playing 'digits/7.gsm' (language 'en') [2015-03-27 09:48:06] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:06] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:06] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:06] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:48:06] DEBUG[16240][C-00000001] say.c: Parsing p (offset 30) in 'vm-received' q 'digits/at' IMp [2015-03-27 09:48:06] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format gsm [2015-03-27 09:48:06] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:48:06] VERBOSE[16240][C-00000001] file.c: Playing 'digits/a-m.gsm' (language 'en') [2015-03-27 09:48:06] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Got RTCP report of 84 bytes [2015-03-27 09:48:06] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:06] DEBUG[16240][C-00000001] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:06] DEBUG[16240][C-00000001] acl.c: Attached to given IP address [2015-03-27 09:48:06] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:06] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:06] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:48:06] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:06] DEBUG[15861] acl.c: Not an IPv4 nor IPv6 address, cannot get port. [2015-03-27 09:48:06] DEBUG[15861] acl.c: Attached to given IP address [2015-03-27 09:48:07] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:07] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:07] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2015-03-27 09:48:07] DEBUG[16240][C-00000001] channel.c: Set channel SIP/testClient-00000001 to write format ulaw [2015-03-27 09:48:07] DEBUG[16240][C-00000001] app_voicemail.c: VM-Duration: duration is: 5 seconds converted to: 0 minutes [2015-03-27 09:48:07] DEBUG[16240][C-00000001] file.c: Ooh, found a video stream, too, format h264 [2015-03-27 09:48:07] DEBUG[16240][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2015-03-27 09:48:07] DEBUG[16240][C-00000001] res_rtp_asterisk.c: Ooh, format changed from none to h264 ... crash ...