cat /var/log/asterisk/myfull [Jun 29 18:18:29] VERBOSE[18577] config.c: == Parsing '/etc/asterisk/logger.conf': [Jun 29 18:18:29] DEBUG[18577] config.c: Parsing /etc/asterisk/logger.conf [Jun 29 18:18:29] VERBOSE[18577] config.c: == Found [Jun 29 18:18:29] VERBOSE[18577] logger.c: Asterisk Queue Logger restarted [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 0 [ 43]: INVITE sip:5555@192.168.20.254:5061 SIP/2.0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK1313155179 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 2 [ 68]: From: "srtp test" ;tag=1859080252 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 3 [ 34]: To: [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 4 [ 34]: Call-ID: 1703354151@192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 6 [ 58]: Contact: [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 8 [115]: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 10 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 11 [ 19]: Supported: replaces [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 12 [ 51]: Allow-Events: talk,hold,conference,refer,check-sync [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 13 [ 19]: Content-Length: 547 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 14 [ 0]: [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: <--- SIP read from TLS:192.168.20.106:5062 ---> INVITE sip:5555@192.168.20.254:5061 SIP/2.0 Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK1313155179 From: "srtp test" ;tag=1859080252 To: Call-ID: 1703354151@192.168.20.106 CSeq: 1 INVITE Contact: Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T26P 6.50.0.50 Supported: replaces Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 547 v=0 o=- 20002 20002 IN IP4 192.168.20.106 s=SDP data c=IN IP4 192.168.20.106 t=0 0 m=audio 11782 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MjcwMjI0ODE1YzBjNGM4MjM1Y2RhOWIzMzY1ZWMx a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NzEwZjQ4NjMxOGRkODNkZGRmZTI2YmYANTgwYzAx a=crypto:3 F8_128_HMAC_SHA1_80 inline:NmMwYzQ1Mzg2MjRmZWU3ZTIyYmU2ODkyOGJlNjUz a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 0 [ 43]: INVITE sip:5555@192.168.20.254:5061 SIP/2.0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK1313155179 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 2 [ 68]: From: "srtp test" ;tag=1859080252 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 3 [ 34]: To: [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 4 [ 34]: Call-ID: 1703354151@192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 6 [ 58]: Contact: [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 8 [115]: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 10 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 11 [ 19]: Supported: replaces [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 12 [ 51]: Allow-Events: talk,hold,conference,refer,check-sync [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 13 [ 19]: Content-Length: 547 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 14 [ 0]: [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 0 [ 3]: v=0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 1 [ 37]: o=- 20002 20002 IN IP4 192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 2 [ 10]: s=SDP data [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 5 [ 35]: m=audio 11782 RTP/SAVP 0 8 18 9 101 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MjcwMjI0ODE1YzBjNGM4MjM1Y2RhOWIzMzY1ZWMx [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 7 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NzEwZjQ4NjMxOGRkODNkZGRmZTI2YmYANTgwYzAx [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 8 [ 78]: a=crypto:3 F8_128_HMAC_SHA1_80 inline:NmMwYzQ1Mzg2MjRmZWU3ZTIyYmU2ODkyOGJlNjUz [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 10 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 11 [ 21]: a=rtpmap:18 G729/8000 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 12 [ 19]: a=fmtp:18 annexb=no [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 13 [ 20]: a=rtpmap:9 G722/8000 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 14 [ 15]: a=fmtp:101 0-15 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 16 [ 10]: a=sendrecv [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: --- (14 headers 17 lines) --- [Jun 29 18:18:36] DEBUG[18565] acl.c: Found IP address for this socket [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Setting SIP_TRANSPORT_TLS with address 192.168.20.254:5061 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Allocating new SIP dialog for 1703354151@192.168.20.106 - INVITE (No RTP) [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Begin: parsing SIP "Supported: replaces" [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Found SIP option: -replaces- [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Matched SIP option: replaces [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Sending to 192.168.20.106 : 5062 (no NAT) [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Initializing initreq for method INVITE - callid 1703354151@192.168.20.106 [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Using INVITE request as basis request - 1703354151@192.168.20.106 [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found peer '123111113' for '123111113' from 192.168.20.106:5062 [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.20.106:5062 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK1313155179;received=192.168.20.106 From: "srtp test" ;tag=1859080252 To: ;tag=as2ae7513e Call-ID: 1703354151@192.168.20.106 CSeq: 1 INVITE Server: Asterisk PBX SVN-trunk-r272447M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="sip.allo.md", nonce="357ba820" Content-Length: 0 <------------> [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Trying to put 'SIP/2.0 401' onto TLS socket destined for 192.168.20.106:5062 [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Scheduling destruction of SIP dialog '1703354151@192.168.20.106' in 6400 ms (Method: INVITE) [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 0 [ 40]: ACK sip:5555@192.168.20.254:5061 SIP/2.0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK1313155179 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 2 [ 68]: From: "srtp test" ;tag=1859080252 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 3 [ 49]: To: ;tag=as2ae7513e [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 4 [ 34]: Call-ID: 1703354151@192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 7 [ 0]: [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: <--- SIP read from TLS:192.168.20.106:5062 ---> ACK sip:5555@192.168.20.254:5061 SIP/2.0 Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK1313155179 From: "srtp test" ;tag=1859080252 To: ;tag=as2ae7513e Call-ID: 1703354151@192.168.20.106 CSeq: 1 ACK Content-Length: 0 <-------------> [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 0 [ 40]: ACK sip:5555@192.168.20.254:5061 SIP/2.0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK1313155179 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 2 [ 68]: From: "srtp test" ;tag=1859080252 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 3 [ 49]: To: ;tag=as2ae7513e [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 4 [ 34]: Call-ID: 1703354151@192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 7 [ 0]: [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: --- (7 headers 0 lines) --- [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Stopping retransmission on '1703354151@192.168.20.106' of Response 1: Match Not Found [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 0 [ 43]: INVITE sip:5555@192.168.20.254:5061 SIP/2.0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK141421712 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 2 [ 68]: From: "srtp test" ;tag=1859080252 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 3 [ 34]: To: [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 4 [ 34]: Call-ID: 1703354151@192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 5 [ 14]: CSeq: 2 INVITE [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 6 [ 58]: Contact: [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 7 [177]: Authorization: Digest username="123111113", realm="sip.allo.md", nonce="357ba820", uri="sip:5555@192.168.20.254:5061", response="3d378e832ecc3e6dcb7411359690770f", algorithm=MD5 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 9 [115]: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 11 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 12 [ 19]: Supported: replaces [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 13 [ 51]: Allow-Events: talk,hold,conference,refer,check-sync [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 14 [ 19]: Content-Length: 547 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 15 [ 0]: [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: <--- SIP read from TLS:192.168.20.106:5062 ---> INVITE sip:5555@192.168.20.254:5061 SIP/2.0 Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK141421712 From: "srtp test" ;tag=1859080252 To: Call-ID: 1703354151@192.168.20.106 CSeq: 2 INVITE Contact: Authorization: Digest username="123111113", realm="sip.allo.md", nonce="357ba820", uri="sip:5555@192.168.20.254:5061", response="3d378e832ecc3e6dcb7411359690770f", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T26P 6.50.0.50 Supported: replaces Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 547 v=0 o=- 20002 20002 IN IP4 192.168.20.106 s=SDP data c=IN IP4 192.168.20.106 t=0 0 m=audio 11782 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MjcwMjI0ODE1YzBjNGM4MjM1Y2RhOWIzMzY1ZWMx a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NzEwZjQ4NjMxOGRkODNkZGRmZTI2YmYANTgwYzAx a=crypto:3 F8_128_HMAC_SHA1_80 inline:NmMwYzQ1Mzg2MjRmZWU3ZTIyYmU2ODkyOGJlNjUz a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 0 [ 43]: INVITE sip:5555@192.168.20.254:5061 SIP/2.0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK141421712 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 2 [ 68]: From: "srtp test" ;tag=1859080252 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 3 [ 34]: To: [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 4 [ 34]: Call-ID: 1703354151@192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 5 [ 14]: CSeq: 2 INVITE [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 6 [ 58]: Contact: [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 7 [177]: Authorization: Digest username="123111113", realm="sip.allo.md", nonce="357ba820", uri="sip:5555@192.168.20.254:5061", response="3d378e832ecc3e6dcb7411359690770f", algorithm=MD5 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 9 [115]: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 11 [ 38]: User-Agent: Yealink SIP-T26P 6.50.0.50 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 12 [ 19]: Supported: replaces [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 13 [ 51]: Allow-Events: talk,hold,conference,refer,check-sync [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 14 [ 19]: Content-Length: 547 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 15 [ 0]: [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 0 [ 3]: v=0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 1 [ 37]: o=- 20002 20002 IN IP4 192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 2 [ 10]: s=SDP data [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 5 [ 35]: m=audio 11782 RTP/SAVP 0 8 18 9 101 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 6 [ 82]: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MjcwMjI0ODE1YzBjNGM4MjM1Y2RhOWIzMzY1ZWMx [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 7 [ 82]: a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NzEwZjQ4NjMxOGRkODNkZGRmZTI2YmYANTgwYzAx [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 8 [ 78]: a=crypto:3 F8_128_HMAC_SHA1_80 inline:NmMwYzQ1Mzg2MjRmZWU3ZTIyYmU2ODkyOGJlNjUz [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 10 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 11 [ 21]: a=rtpmap:18 G729/8000 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 12 [ 19]: a=fmtp:18 annexb=no [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 13 [ 20]: a=rtpmap:9 G722/8000 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 14 [ 15]: a=fmtp:101 0-15 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Body 16 [ 10]: a=sendrecv [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: --- (15 headers 17 lines) --- [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Sending to 192.168.20.106 : 5062 (no NAT) [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Initializing initreq for method INVITE - callid 1703354151@192.168.20.106 [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Using INVITE request as basis request - 1703354151@192.168.20.106 [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found peer '123111113' for '123111113' from 192.168.20.106:5062 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9683118' [Jun 29 18:18:36] DEBUG[18565] res_rtp_asterisk.c: Allocated port 12658 for RTP instance '0x9683118' [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: RTP instance '0x9683118' is setup and ready to go [Jun 29 18:18:36] DEBUG[18565] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9683118' [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Setting NAT on RTP to Off [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing session-level SDP o=- 20002 20002 IN IP4 192.168.20.106... UNSUPPORTED. [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing session-level SDP s=SDP data... UNSUPPORTED. [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.20.106... OK. [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found RTP audio format 0 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Setting payload 0 based on m type on 0xb12ecca4 [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found RTP audio format 8 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Setting payload 8 based on m type on 0xb12ecca4 [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found RTP audio format 18 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Setting payload 18 based on m type on 0xb12ecca4 [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found RTP audio format 9 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Setting payload 9 based on m type on 0xb12ecca4 [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found RTP audio format 101 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Setting payload 101 based on m type on 0xb12ecca4 [Jun 29 18:18:36] DEBUG[18565] sip/sdp_crypto.c: local_key64 KQKhrzrJ9mCWAD29BpvaUu5UWrnvsl9Ur1zAH04t len 40 [Jun 29 18:18:36] DEBUG[18565] sip/sdp_crypto.c: SRTP policy activated [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MjcwMjI0ODE1YzBjNGM4MjM1Y2RhOWIzMzY1ZWMx... OK. [Jun 29 18:18:36] DEBUG[18565] sip/sdp_crypto.c: SRTP policy activated [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NzEwZjQ4NjMxOGRkODNkZGRmZTI2YmYANTgwYzAx... OK. [Jun 29 18:18:36] WARNING[18565] sip/sdp_crypto.c: Unsupported crypto suite: F8_128_HMAC_SHA1_80 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=crypto:3 F8_128_HMAC_SHA1_80 inline:NmMwYzQ1Mzg2MjRmZWU3ZTIyYmU2ODkyOGJlNjUz... UNSUPPORTED. [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found audio description format PCMU for ID 0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found audio description format G729 for ID 18 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found audio description format G722 for ID 9 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Incorporating payload 0 on 0xb12ecca4 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Incorporating payload 8 on 0xb12ecca4 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Incorporating payload 9 on 0xb12ecca4 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Incorporating payload 18 on 0xb12ecca4 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Incorporating payload 101 on 0xb12ecca4 [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 29 18:18:36] DEBUG[18565] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9683118' [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Peer audio RTP is at port 192.168.20.106:11782 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Copying payload 0 from 0xb12ecca4 to 0x9683168 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Copying payload 8 from 0xb12ecca4 to 0x9683168 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Copying payload 9 from 0xb12ecca4 to 0x9683168 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Copying payload 18 from 0xb12ecca4 to 0x9683168 [Jun 29 18:18:36] DEBUG[18565] rtp_engine.c: Copying payload 101 from 0xb12ecca4 to 0x9683168 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Checking SIP call limits for device 123111113 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Updating call counter for incoming call [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: Looking for 5555 in from-internal (domain 192.168.20.254) [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: This channel will not be able to handle video. [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: build_route: Contact hop: [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: list_route: hop: [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: SIP/123111113-00000003: New call is still down.... Trying... [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: <--- Transmitting (no NAT) to 192.168.20.106:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK141421712;received=192.168.20.106 From: "srtp test" ;tag=1859080252 To: Call-ID: 1703354151@192.168.20.106 CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r272447M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Trying to put 'SIP/2.0 100' onto TLS socket destined for 192.168.20.106:5062 [Jun 29 18:18:36] DEBUG[18578] pbx.c: Launching 'NoOp' [Jun 29 18:18:36] VERBOSE[18578] pbx.c: -- Executing [5555@from-internal:1] NoOp("SIP/123111113-00000003", "TEST ") in new stack [Jun 29 18:18:36] DEBUG[18501] devicestate.c: No provider found, checking channel drivers for SIP - 123111113 [Jun 29 18:18:36] DEBUG[18578] pbx.c: Result of 'EXTEN' is '5555' [Jun 29 18:18:36] DEBUG[18501] chan_sip.c: Checking device state for peer 123111113 [Jun 29 18:18:36] DEBUG[18578] pbx.c: Launching 'Dial' [Jun 29 18:18:36] DEBUG[18501] devicestate.c: Changing state for SIP/123111113 - state 1 (Not in use) [Jun 29 18:18:36] VERBOSE[18578] pbx.c: -- Executing [5555@from-internal:2] Dial("SIP/123111113-00000003", "DAHDI/g1/5555,,trTR") in new stack [Jun 29 18:18:36] DEBUG[18501] devicestate.c: device 'SIP/123111113' state '1' [Jun 29 18:18:36] WARNING[18578] channel.c: No channel type registered for 'DAHDI' [Jun 29 18:18:36] WARNING[18578] app_dial.c: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented) [Jun 29 18:18:36] VERBOSE[18578] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1) [Jun 29 18:18:36] DEBUG[18578] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. [Jun 29 18:18:36] DEBUG[18578] pbx.c: Launching 'Hangup' [Jun 29 18:18:36] VERBOSE[18578] pbx.c: -- Executing [5555@from-internal:3] Hangup("SIP/123111113-00000003", "") in new stack [Jun 29 18:18:36] DEBUG[18578] pbx.c: Spawn extension (from-internal,5555,3) exited non-zero on 'SIP/123111113-00000003' [Jun 29 18:18:36] VERBOSE[18578] pbx.c: == Spawn extension (from-internal, 5555, 3) exited non-zero on 'SIP/123111113-00000003' [Jun 29 18:18:36] DEBUG[18578] channel.c: Soft-Hanging up channel 'SIP/123111113-00000003' [Jun 29 18:18:36] DEBUG[18578] channel.c: Hanging up channel 'SIP/123111113-00000003' [Jun 29 18:18:36] DEBUG[18578] chan_sip.c: Hangup call SIP/123111113-00000003, SIP callid 1703354151@192.168.20.106 [Jun 29 18:18:36] DEBUG[18578] chan_sip.c: Hanging up channel in state Ring (not UP) [Jun 29 18:18:36] DEBUG[18578] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9683118' [Jun 29 18:18:36] VERBOSE[18578] chan_sip.c: Scheduling destruction of SIP dialog '1703354151@192.168.20.106' in 6400 ms (Method: INVITE) [Jun 29 18:18:36] VERBOSE[18578] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.20.106:5062 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK141421712;received=192.168.20.106 From: "srtp test" ;tag=1859080252 To: ;tag=as3f4ec719 Call-ID: 1703354151@192.168.20.106 CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r272447M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jun 29 18:18:36] DEBUG[18578] chan_sip.c: Trying to put 'SIP/2.0 503' onto TLS socket destined for 192.168.20.106:5062 [Jun 29 18:18:36] DEBUG[18501] devicestate.c: No provider found, checking channel drivers for SIP - 123111113 [Jun 29 18:18:36] DEBUG[18501] chan_sip.c: Checking device state for peer 123111113 [Jun 29 18:18:36] DEBUG[18501] devicestate.c: Changing state for SIP/123111113 - state 1 (Not in use) [Jun 29 18:18:36] DEBUG[18501] devicestate.c: device 'SIP/123111113' state '1' [Jun 29 18:18:36] DEBUG[18509] app_queue.c: Device 'SIP/123111113' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 29 18:18:36] DEBUG[18509] app_queue.c: Device 'SIP/123111113' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 0 [ 40]: ACK sip:5555@192.168.20.254:5061 SIP/2.0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK141421712 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 2 [ 68]: From: "srtp test" ;tag=1859080252 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 3 [ 49]: To: ;tag=as3f4ec719 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 4 [ 34]: Call-ID: 1703354151@192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 5 [ 11]: CSeq: 2 ACK [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 7 [ 0]: [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: <--- SIP read from TLS:192.168.20.106:5062 ---> ACK sip:5555@192.168.20.254:5061 SIP/2.0 Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK141421712 From: "srtp test" ;tag=1859080252 To: ;tag=as3f4ec719 Call-ID: 1703354151@192.168.20.106 CSeq: 2 ACK Content-Length: 0 <-------------> [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 0 [ 40]: ACK sip:5555@192.168.20.254:5061 SIP/2.0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 1 [ 60]: Via: SIP/2.0/TLS 192.168.20.106:5062;branch=z9hG4bK141421712 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 2 [ 68]: From: "srtp test" ;tag=1859080252 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 3 [ 49]: To: ;tag=as3f4ec719 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 4 [ 34]: Call-ID: 1703354151@192.168.20.106 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 5 [ 11]: CSeq: 2 ACK [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Header 7 [ 0]: [Jun 29 18:18:36] VERBOSE[18565] chan_sip.c: --- (7 headers 0 lines) --- [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 29 18:18:36] DEBUG[18565] chan_sip.c: Stopping retransmission on '1703354151@192.168.20.106' of Response 2: Match Not Found [Jun 29 18:18:42] DEBUG[18512] chan_sip.c: Auto destroying SIP dialog '1703354151@192.168.20.106' [Jun 29 18:18:42] DEBUG[18512] chan_sip.c: Destroying SIP dialog 1703354151@192.168.20.106 [Jun 29 18:18:42] VERBOSE[18512] chan_sip.c: Really destroying SIP dialog '1703354151@192.168.20.106' Method: INVITE [Jun 29 18:18:42] DEBUG[18512] rtp_engine.c: Destroyed RTP instance '0x9683118' [Jun 29 18:18:44] VERBOSE[18577] asterisk.c: Beginning asterisk shutdown.... [Jun 29 18:18:44] VERBOSE[18577] asterisk.c: Executing last minute cleanups [Jun 29 18:18:44] VERBOSE[18577] res_musiconhold.c: == Destroying musiconhold processes [Jun 29 18:18:44] DEBUG[18577] res_musiconhold.c: Destroying MOH class 'default' [Jun 29 18:18:44] VERBOSE[18577] asterisk.c: Asterisk cleanly ending (0).