CLI> sip set debug on SIP Debugging enabled *CLI> [Nov 9 18:20:27] [Nov 9 18:20:27] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 9 18:20:27] INVITE sip:6100@192.168.210.132 SIP/2.0 [Nov 9 18:20:27] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00e906790ea5e61184fe5c199ea0d207;rport [Nov 9 18:20:27] From: "PhonerLite" ;tag=859153178 [Nov 9 18:20:27] To: [Nov 9 18:20:27] Call-ID: 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:27] CSeq: 136 INVITE [Nov 9 18:20:27] Contact: [Nov 9 18:20:27] Content-Type: application/sdp [Nov 9 18:20:27] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Nov 9 18:20:27] Max-Forwards: 70 [Nov 9 18:20:27] Supported: 100rel, replaces, from-change [Nov 9 18:20:27] P-Early-Media: supported [Nov 9 18:20:27] User-Agent: SIPPER for PhonerLite [Nov 9 18:20:27] P-Preferred-Identity: [Nov 9 18:20:27] Content-Length: 451 [Nov 9 18:20:27] [Nov 9 18:20:27] v=0 [Nov 9 18:20:27] o=- 3119471676 1 IN IP4 192.168.210.111 [Nov 9 18:20:27] s=SIPPER for PhonerLite [Nov 9 18:20:27] c=IN IP4 192.168.210.111 [Nov 9 18:20:27] t=0 0 [Nov 9 18:20:27] m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 101 [Nov 9 18:20:27] a=rtpmap:107 opus/48000/2 [Nov 9 18:20:27] a=rtpmap:8 PCMA/8000 [Nov 9 18:20:27] a=rtpmap:0 PCMU/8000 [Nov 9 18:20:27] a=rtpmap:2 G726-32/8000 [Nov 9 18:20:27] a=rtpmap:3 GSM/8000 [Nov 9 18:20:27] a=rtpmap:97 iLBC/8000 [Nov 9 18:20:27] a=rtpmap:110 speex/8000 [Nov 9 18:20:27] a=rtpmap:111 speex/16000 [Nov 9 18:20:27] a=rtpmap:9 G722/8000 [Nov 9 18:20:27] a=rtpmap:101 telephone-event/8000 [Nov 9 18:20:27] a=fmtp:101 0-16 [Nov 9 18:20:27] a=ssrc:2412325951 [Nov 9 18:20:27] a=sendrecv [Nov 9 18:20:27] <-------------> [Nov 9 18:20:27] --- (15 headers 19 lines) --- [Nov 9 18:20:27] Sending to 192.168.210.111:5060 (no NAT) [Nov 9 18:20:27] Sending to 192.168.210.111:5060 (no NAT) [Nov 9 18:20:27] Using INVITE request as basis request - 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:27] Found peer '290' for '290' from 192.168.210.111:5060 [Nov 9 18:20:27] [Nov 9 18:20:27] <--- Reliably Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 9 18:20:27] SIP/2.0 401 Unauthorized [Nov 9 18:20:27] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00e906790ea5e61184fe5c199ea0d207;received=192.168.210.111;rport=5060 [Nov 9 18:20:27] From: "PhonerLite" ;tag=859153178 [Nov 9 18:20:27] To: ;tag=as3474224c [Nov 9 18:20:27] Call-ID: 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:27] CSeq: 136 INVITE [Nov 9 18:20:27] Server: Asterisk PBX GIT-master-0d85f18 [Nov 9 18:20:27] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 9 18:20:27] Supported: replaces, timer [Nov 9 18:20:27] WWW-Authenticate: Digest algorithm=MD5, realm="pc_dany_asterisk", nonce="42ad9e32" [Nov 9 18:20:27] Content-Length: 0 [Nov 9 18:20:27] [Nov 9 18:20:27] [Nov 9 18:20:27] <------------> [Nov 9 18:20:27] Scheduling destruction of SIP dialog '00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111' in 32000 ms (Method: INVITE) [Nov 9 18:20:27] [Nov 9 18:20:27] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 9 18:20:27] ACK sip:6100@192.168.210.132 SIP/2.0 [Nov 9 18:20:27] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00e906790ea5e61184fe5c199ea0d207;rport [Nov 9 18:20:27] From: "PhonerLite" ;tag=859153178 [Nov 9 18:20:27] To: ;tag=as3474224c [Nov 9 18:20:27] Call-ID: 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:27] CSeq: 136 ACK [Nov 9 18:20:27] Content-Length: 0 [Nov 9 18:20:27] [Nov 9 18:20:27] <-------------> [Nov 9 18:20:27] --- (7 headers 0 lines) --- [Nov 9 18:20:27] [Nov 9 18:20:27] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 9 18:20:27] INVITE sip:6100@192.168.210.132 SIP/2.0 [Nov 9 18:20:27] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00e906790ea5e61184ff5c199ea0d207;rport [Nov 9 18:20:27] From: "PhonerLite" ;tag=859153178 [Nov 9 18:20:27] To: [Nov 9 18:20:27] Call-ID: 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:27] CSeq: 137 INVITE [Nov 9 18:20:27] Contact: [Nov 9 18:20:27] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="42ad9e32", uri="sip:6100@192.168.210.132", response="dbf366d31ec279a14b102a6d4ef34b28", algorithm=MD5 [Nov 9 18:20:27] Content-Type: application/sdp [Nov 9 18:20:27] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Nov 9 18:20:27] Max-Forwards: 70 [Nov 9 18:20:27] Supported: 100rel, replaces, from-change [Nov 9 18:20:27] P-Early-Media: supported [Nov 9 18:20:27] User-Agent: SIPPER for PhonerLite [Nov 9 18:20:27] P-Preferred-Identity: [Nov 9 18:20:27] Content-Length: 451 [Nov 9 18:20:27] [Nov 9 18:20:27] v=0 [Nov 9 18:20:27] o=- 3119471676 1 IN IP4 192.168.210.111 [Nov 9 18:20:27] s=SIPPER for PhonerLite [Nov 9 18:20:27] c=IN IP4 192.168.210.111 [Nov 9 18:20:27] t=0 0 [Nov 9 18:20:27] m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 101 [Nov 9 18:20:27] a=rtpmap:107 opus/48000/2 [Nov 9 18:20:27] a=rtpmap:8 PCMA/8000 [Nov 9 18:20:27] a=rtpmap:0 PCMU/8000 [Nov 9 18:20:27] a=rtpmap:2 G726-32/8000 [Nov 9 18:20:27] a=rtpmap:3 GSM/8000 [Nov 9 18:20:27] a=rtpmap:97 iLBC/8000 [Nov 9 18:20:27] a=rtpmap:110 speex/8000 [Nov 9 18:20:27] a=rtpmap:111 speex/16000 [Nov 9 18:20:27] a=rtpmap:9 G722/8000 [Nov 9 18:20:27] a=rtpmap:101 telephone-event/8000 [Nov 9 18:20:27] a=fmtp:101 0-16 [Nov 9 18:20:27] a=ssrc:2412325951 [Nov 9 18:20:27] a=sendrecv [Nov 9 18:20:27] <-------------> [Nov 9 18:20:27] --- (16 headers 19 lines) --- [Nov 9 18:20:27] Sending to 192.168.210.111:5060 (no NAT) [Nov 9 18:20:27] Using INVITE request as basis request - 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:27] Found peer '290' for '290' from 192.168.210.111:5060 [Nov 9 18:20:27] == Using SIP RTP CoS mark 5 [Nov 9 18:20:27] Found RTP audio format 107 [Nov 9 18:20:27] Found RTP audio format 8 [Nov 9 18:20:27] Found RTP audio format 0 [Nov 9 18:20:27] Found RTP audio format 2 [Nov 9 18:20:27] Found RTP audio format 3 [Nov 9 18:20:27] Found RTP audio format 97 [Nov 9 18:20:27] Found RTP audio format 110 [Nov 9 18:20:27] Found RTP audio format 111 [Nov 9 18:20:27] Found RTP audio format 9 [Nov 9 18:20:27] Found RTP audio format 101 [Nov 9 18:20:27] Found audio description format opus for ID 107 [Nov 9 18:20:27] Found audio description format PCMA for ID 8 [Nov 9 18:20:27] Found audio description format PCMU for ID 0 [Nov 9 18:20:27] Found audio description format G726-32 for ID 2 [Nov 9 18:20:27] Found audio description format GSM for ID 3 [Nov 9 18:20:27] Found audio description format iLBC for ID 97 [Nov 9 18:20:27] Found audio description format speex for ID 110 [Nov 9 18:20:27] Found audio description format speex for ID 111 [Nov 9 18:20:27] Found audio description format G722 for ID 9 [Nov 9 18:20:27] Found audio description format telephone-event for ID 101 [Nov 9 18:20:27] Capabilities: us - (alaw), peer - audio=(ulaw|gsm|alaw|g722|ilbc|opus|speex|speex16)/video=(nothing)/text=(nothing), combined - (alaw) [Nov 9 18:20:27] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 9 18:20:27] Peer audio RTP is at port 192.168.210.111:5062 [Nov 9 18:20:27] Peer doesn't provide T.140 [Nov 9 18:20:27] Looking for 6100 in LocalSets (domain 192.168.210.132) [Nov 9 18:20:27] sip_route_dump: route/path hop: [Nov 9 18:20:27] [Nov 9 18:20:27] <--- Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 9 18:20:27] SIP/2.0 100 Trying [Nov 9 18:20:27] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00e906790ea5e61184ff5c199ea0d207;received=192.168.210.111;rport=5060 [Nov 9 18:20:27] From: "PhonerLite" ;tag=859153178 [Nov 9 18:20:27] To: [Nov 9 18:20:27] Call-ID: 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:27] CSeq: 137 INVITE [Nov 9 18:20:27] Server: Asterisk PBX GIT-master-0d85f18 [Nov 9 18:20:27] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 9 18:20:27] Supported: replaces, timer [Nov 9 18:20:27] Contact: [Nov 9 18:20:27] Content-Length: 0 [Nov 9 18:20:27] [Nov 9 18:20:27] [Nov 9 18:20:27] <------------> [Nov 9 18:20:27] -- Executing [6100@LocalSets:1] NoOp("SIP/290-0000000a", "") in new stack [Nov 9 18:20:27] -- Executing [6100@LocalSets:2] Answer("SIP/290-0000000a", "") in new stack [Nov 9 18:20:27] Audio is at 13336 [Nov 9 18:20:27] Adding codec alaw to SDP [Nov 9 18:20:27] Adding non-codec 0x1 (telephone-event) to SDP [Nov 9 18:20:27] [Nov 9 18:20:27] <--- Reliably Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 9 18:20:27] SIP/2.0 200 OK [Nov 9 18:20:27] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00e906790ea5e61184ff5c199ea0d207;received=192.168.210.111;rport=5060 [Nov 9 18:20:27] From: "PhonerLite" ;tag=859153178 [Nov 9 18:20:27] To: ;tag=as4740f019 [Nov 9 18:20:27] Call-ID: 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:27] CSeq: 137 INVITE [Nov 9 18:20:27] Server: Asterisk PBX GIT-master-0d85f18 [Nov 9 18:20:27] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 9 18:20:27] Supported: replaces, timer [Nov 9 18:20:27] Contact: [Nov 9 18:20:27] Content-Type: application/sdp [Nov 9 18:20:27] Content-Length: 257 [Nov 9 18:20:27] [Nov 9 18:20:27] v=0 [Nov 9 18:20:27] o=root 1229992908 1229992908 IN IP4 192.168.210.132 [Nov 9 18:20:27] s=Asterisk PBX GIT-master-0d85f18 [Nov 9 18:20:27] c=IN IP4 192.168.210.132 [Nov 9 18:20:27] t=0 0 [Nov 9 18:20:27] m=audio 13336 RTP/AVP 8 101 [Nov 9 18:20:27] a=rtpmap:8 PCMA/8000 [Nov 9 18:20:27] a=rtpmap:101 telephone-event/8000 [Nov 9 18:20:27] a=fmtp:101 0-16 [Nov 9 18:20:27] a=maxptime:150 [Nov 9 18:20:27] a=sendrecv [Nov 9 18:20:27] [Nov 9 18:20:27] <------------> [Nov 9 18:20:27] [Nov 9 18:20:27] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 9 18:20:27] ACK sip:6100@192.168.210.132:5060 SIP/2.0 [Nov 9 18:20:27] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00e906790ea5e61185005c199ea0d207;rport [Nov 9 18:20:27] From: "PhonerLite" ;tag=859153178 [Nov 9 18:20:27] To: ;tag=as4740f019 [Nov 9 18:20:27] Call-ID: 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:27] CSeq: 137 ACK [Nov 9 18:20:27] Contact: [Nov 9 18:20:27] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="42ad9e32", uri="sip:6100@192.168.210.132:5060", response="41cd82cf0f4320558b00548405fde8b4", algorithm=MD5 [Nov 9 18:20:27] Max-Forwards: 70 [Nov 9 18:20:27] Content-Length: 0 [Nov 9 18:20:27] [Nov 9 18:20:27] <-------------> [Nov 9 18:20:27] --- (10 headers 0 lines) --- [Nov 9 18:20:27] > 0xb6920930 -- Probation passed - setting RTP source address to 192.168.210.111:5062 [Nov 9 18:20:27] -- Executing [6100@LocalSets:3] Stasis("SIP/290-0000000a", "attendant") in new stack [Nov 9 18:20:27] Really destroying SIP dialog '80AF8C66-0EA5-E611-84FA-5C199EA0D207@192.168.210.111' Method: ACK [Nov 9 18:20:27] Really destroying SIP dialog '00BF9163-0EA5-E611-84F7-5C199EA0D207@192.168.210.111' Method: BYE [Nov 9 18:20:30] [Nov 9 18:20:30] <--- SIP read from UDP:192.168.210.71:5060 ---> [Nov 9 18:20:30] REGISTER sip:192.168.210.132 SIP/2.0 [Nov 9 18:20:30] Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK60b2142a [Nov 9 18:20:30] Max-Forwards: 70 [Nov 9 18:20:30] From: ;tag=as316218a5 [Nov 9 18:20:30] To: [Nov 9 18:20:30] Call-ID: 570cd08e593453b70f717b397c0048bc@127.0.1.1 [Nov 9 18:20:30] CSeq: 232 REGISTER [Nov 9 18:20:30] Supported: replaces, timer [Nov 9 18:20:30] User-Agent: Asterisk PBX 14.0.2 [Nov 9 18:20:30] Authorization: Digest username="toronto", realm="pc_dany_asterisk", algorithm=MD5, uri="sip:192.168.210.132", nonce="464a4763", response="62238b521555f8d2e2b8eb07c865158b" [Nov 9 18:20:30] Expires: 120 [Nov 9 18:20:30] Contact: [Nov 9 18:20:30] Content-Length: 0 [Nov 9 18:20:30] [Nov 9 18:20:30] <-------------> [Nov 9 18:20:30] --- (13 headers 0 lines) --- [Nov 9 18:20:30] Sending to 192.168.210.71:5060 (no NAT) [Nov 9 18:20:30] Sending to 192.168.210.71:5060 (no NAT) [Nov 9 18:20:30] [Nov 9 18:20:30] <--- Transmitting (no NAT) to 192.168.210.71:5060 ---> [Nov 9 18:20:30] SIP/2.0 401 Unauthorized [Nov 9 18:20:30] Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK60b2142a;received=192.168.210.71 [Nov 9 18:20:30] From: ;tag=as316218a5 [Nov 9 18:20:30] To: ;tag=as69927b22 [Nov 9 18:20:30] Call-ID: 570cd08e593453b70f717b397c0048bc@127.0.1.1 [Nov 9 18:20:30] CSeq: 232 REGISTER [Nov 9 18:20:30] Server: Asterisk PBX GIT-master-0d85f18 [Nov 9 18:20:30] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 9 18:20:30] Supported: replaces, timer [Nov 9 18:20:30] WWW-Authenticate: Digest algorithm=MD5, realm="pc_dany_asterisk", nonce="6b1ba3bc" [Nov 9 18:20:30] Content-Length: 0 [Nov 9 18:20:30] [Nov 9 18:20:30] [Nov 9 18:20:30] <------------> [Nov 9 18:20:30] Scheduling destruction of SIP dialog '570cd08e593453b70f717b397c0048bc@127.0.1.1' in 32000 ms (Method: REGISTER) [Nov 9 18:20:30] [Nov 9 18:20:30] <--- SIP read from UDP:192.168.210.71:5060 ---> [Nov 9 18:20:30] REGISTER sip:192.168.210.132 SIP/2.0 [Nov 9 18:20:30] Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK3232627d [Nov 9 18:20:30] Max-Forwards: 70 [Nov 9 18:20:30] From: ;tag=as316218a5 [Nov 9 18:20:30] To: [Nov 9 18:20:30] Call-ID: 570cd08e593453b70f717b397c0048bc@127.0.1.1 [Nov 9 18:20:30] CSeq: 233 REGISTER [Nov 9 18:20:30] Supported: replaces, timer [Nov 9 18:20:30] User-Agent: Asterisk PBX 14.0.2 [Nov 9 18:20:30] Authorization: Digest username="toronto", realm="pc_dany_asterisk", algorithm=MD5, uri="sip:192.168.210.132", nonce="6b1ba3bc", response="679a880cb3c8c0af969f60b86041b2dc" [Nov 9 18:20:30] Expires: 120 [Nov 9 18:20:30] Contact: [Nov 9 18:20:30] Content-Length: 0 [Nov 9 18:20:30] [Nov 9 18:20:30] <-------------> [Nov 9 18:20:30] --- (13 headers 0 lines) --- [Nov 9 18:20:30] Sending to 192.168.210.71:5060 (no NAT) [Nov 9 18:20:30] [Nov 9 18:20:30] <--- Transmitting (no NAT) to 192.168.210.71:5060 ---> [Nov 9 18:20:30] SIP/2.0 200 OK [Nov 9 18:20:30] Via: SIP/2.0/UDP 192.168.210.71:5060;branch=z9hG4bK3232627d;received=192.168.210.71 [Nov 9 18:20:30] From: ;tag=as316218a5 [Nov 9 18:20:30] To: ;tag=as69927b22 [Nov 9 18:20:30] Call-ID: 570cd08e593453b70f717b397c0048bc@127.0.1.1 [Nov 9 18:20:30] CSeq: 233 REGISTER [Nov 9 18:20:30] Server: Asterisk PBX GIT-master-0d85f18 [Nov 9 18:20:30] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 9 18:20:30] Supported: replaces, timer [Nov 9 18:20:30] Expires: 120 [Nov 9 18:20:30] Contact: ;expires=120 [Nov 9 18:20:30] Date: Wed, 09 Nov 2016 17:20:30 GMT [Nov 9 18:20:30] Content-Length: 0 [Nov 9 18:20:30] [Nov 9 18:20:30] [Nov 9 18:20:30] <------------> [Nov 9 18:20:30] Scheduling destruction of SIP dialog '570cd08e593453b70f717b397c0048bc@127.0.1.1' in 32000 ms (Method: REGISTER) [Nov 9 18:20:31] set_destination: Parsing for address/port to send to [Nov 9 18:20:31] set_destination: set destination to 192.168.210.111:5060 [Nov 9 18:20:31] Reliably Transmitting (no NAT) to 192.168.210.111:5060: [Nov 9 18:20:31] REFER sip:290@192.168.210.111:5060 SIP/2.0 [Nov 9 18:20:31] Via: SIP/2.0/UDP 192.168.210.132:5060;branch=z9hG4bK69c83b38;rport [Nov 9 18:20:31] Max-Forwards: 70 [Nov 9 18:20:31] From: ;tag=as4740f019 [Nov 9 18:20:31] To: "PhonerLite" ;tag=859153178 [Nov 9 18:20:31] Contact: [Nov 9 18:20:31] Call-ID: 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] CSeq: 102 REFER [Nov 9 18:20:31] User-Agent: Asterisk PBX GIT-master-0d85f18 [Nov 9 18:20:31] Date: Wed, 09 Nov 2016 17:20:31 GMT [Nov 9 18:20:31] Refer-To: [Nov 9 18:20:31] Referred-By: [Nov 9 18:20:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 9 18:20:31] Supported: replaces, timer [Nov 9 18:20:31] Content-Length: 0 [Nov 9 18:20:31] [Nov 9 18:20:31] [Nov 9 18:20:31] --- [Nov 9 18:20:31] [Nov 9 18:20:31] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 9 18:20:31] SIP/2.0 202 Accepted [Nov 9 18:20:31] Via: SIP/2.0/UDP 192.168.210.132:5060;branch=z9hG4bK69c83b38;rport=5060 [Nov 9 18:20:31] From: ;tag=as4740f019 [Nov 9 18:20:31] To: "PhonerLite" ;tag=859153178 [Nov 9 18:20:31] Call-ID: 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] CSeq: 102 REFER [Nov 9 18:20:31] Contact: [Nov 9 18:20:31] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Nov 9 18:20:31] Server: SIPPER for PhonerLite [Nov 9 18:20:31] Content-Length: 0 [Nov 9 18:20:31] [Nov 9 18:20:31] <-------------> [Nov 9 18:20:31] --- (10 headers 0 lines) --- [Nov 9 18:20:31] [Nov 9 18:20:31] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 9 18:20:31] INVITE sip:toronto101@192.168.210.132 SIP/2.0 [Nov 9 18:20:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80d9017c0ea5e61185015c199ea0d207;rport [Nov 9 18:20:31] From: "PhonerLite" ;tag=1111206135 [Nov 9 18:20:31] To: [Nov 9 18:20:31] Call-ID: 80D9017C-0EA5-E611-8500-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] CSeq: 138 INVITE [Nov 9 18:20:31] Contact: [Nov 9 18:20:31] Content-Type: application/sdp [Nov 9 18:20:31] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Nov 9 18:20:31] Max-Forwards: 70 [Nov 9 18:20:31] Supported: 100rel, replaces, from-change [Nov 9 18:20:31] P-Early-Media: supported [Nov 9 18:20:31] User-Agent: SIPPER for PhonerLite [Nov 9 18:20:31] P-Preferred-Identity: [Nov 9 18:20:31] Content-Length: 450 [Nov 9 18:20:31] [Nov 9 18:20:31] v=0 [Nov 9 18:20:31] o=- 771847758 1 IN IP4 192.168.210.111 [Nov 9 18:20:31] s=SIPPER for PhonerLite [Nov 9 18:20:31] c=IN IP4 192.168.210.111 [Nov 9 18:20:31] t=0 0 [Nov 9 18:20:31] m=audio 5064 RTP/AVP 107 8 0 2 3 97 110 111 9 101 [Nov 9 18:20:31] a=rtpmap:107 opus/48000/2 [Nov 9 18:20:31] a=rtpmap:8 PCMA/8000 [Nov 9 18:20:31] a=rtpmap:0 PCMU/8000 [Nov 9 18:20:31] a=rtpmap:2 G726-32/8000 [Nov 9 18:20:31] a=rtpmap:3 GSM/8000 [Nov 9 18:20:31] a=rtpmap:97 iLBC/8000 [Nov 9 18:20:31] a=rtpmap:110 speex/8000 [Nov 9 18:20:31] a=rtpmap:111 speex/16000 [Nov 9 18:20:31] a=rtpmap:9 G722/8000 [Nov 9 18:20:31] a=rtpmap:101 telephone-event/8000 [Nov 9 18:20:31] a=fmtp:101 0-16 [Nov 9 18:20:31] a=ssrc:2516751377 [Nov 9 18:20:31] a=sendrecv [Nov 9 18:20:31] <-------------> [Nov 9 18:20:31] --- (15 headers 19 lines) --- [Nov 9 18:20:31] Sending to 192.168.210.111:5060 (no NAT) [Nov 9 18:20:31] Sending to 192.168.210.111:5060 (no NAT) [Nov 9 18:20:31] Using INVITE request as basis request - 80D9017C-0EA5-E611-8500-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] Found peer '290' for '290' from 192.168.210.111:5060 [Nov 9 18:20:31] [Nov 9 18:20:31] <--- Reliably Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 9 18:20:31] SIP/2.0 401 Unauthorized [Nov 9 18:20:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80d9017c0ea5e61185015c199ea0d207;received=192.168.210.111;rport=5060 [Nov 9 18:20:31] From: "PhonerLite" ;tag=1111206135 [Nov 9 18:20:31] To: ;tag=as5ee80744 [Nov 9 18:20:31] Call-ID: 80D9017C-0EA5-E611-8500-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] CSeq: 138 INVITE [Nov 9 18:20:31] Server: Asterisk PBX GIT-master-0d85f18 [Nov 9 18:20:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 9 18:20:31] Supported: replaces, timer [Nov 9 18:20:31] WWW-Authenticate: Digest algorithm=MD5, realm="pc_dany_asterisk", nonce="4a9d526e" [Nov 9 18:20:31] Content-Length: 0 [Nov 9 18:20:31] [Nov 9 18:20:31] [Nov 9 18:20:31] <------------> [Nov 9 18:20:31] Scheduling destruction of SIP dialog '80D9017C-0EA5-E611-8500-5C199EA0D207@192.168.210.111' in 32000 ms (Method: INVITE) [Nov 9 18:20:31] [Nov 9 18:20:31] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 9 18:20:31] ACK sip:toronto101@192.168.210.132 SIP/2.0 [Nov 9 18:20:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80d9017c0ea5e61185015c199ea0d207;rport [Nov 9 18:20:31] From: "PhonerLite" ;tag=1111206135 [Nov 9 18:20:31] To: ;tag=as5ee80744 [Nov 9 18:20:31] Call-ID: 80D9017C-0EA5-E611-8500-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] CSeq: 138 ACK [Nov 9 18:20:31] Content-Length: 0 [Nov 9 18:20:31] [Nov 9 18:20:31] <-------------> [Nov 9 18:20:31] --- (7 headers 0 lines) --- [Nov 9 18:20:31] [Nov 9 18:20:31] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 9 18:20:31] INVITE sip:toronto101@192.168.210.132 SIP/2.0 [Nov 9 18:20:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80d9017c0ea5e61185025c199ea0d207;rport [Nov 9 18:20:31] From: "PhonerLite" ;tag=1111206135 [Nov 9 18:20:31] To: [Nov 9 18:20:31] Call-ID: 80D9017C-0EA5-E611-8500-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] CSeq: 139 INVITE [Nov 9 18:20:31] Contact: [Nov 9 18:20:31] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="4a9d526e", uri="sip:toronto101@192.168.210.132", response="813c7e002a55294dbf52bed8e938834f", algorithm=MD5 [Nov 9 18:20:31] Content-Type: application/sdp [Nov 9 18:20:31] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Nov 9 18:20:31] Max-Forwards: 70 [Nov 9 18:20:31] Supported: 100rel, replaces, from-change [Nov 9 18:20:31] P-Early-Media: supported [Nov 9 18:20:31] User-Agent: SIPPER for PhonerLite [Nov 9 18:20:31] P-Preferred-Identity: [Nov 9 18:20:31] Content-Length: 450 [Nov 9 18:20:31] [Nov 9 18:20:31] v=0 [Nov 9 18:20:31] o=- 771847758 1 IN IP4 192.168.210.111 [Nov 9 18:20:31] s=SIPPER for PhonerLite [Nov 9 18:20:31] c=IN IP4 192.168.210.111 [Nov 9 18:20:31] t=0 0 [Nov 9 18:20:31] m=audio 5064 RTP/AVP 107 8 0 2 3 97 110 111 9 101 [Nov 9 18:20:31] a=rtpmap:107 opus/48000/2 [Nov 9 18:20:31] a=rtpmap:8 PCMA/8000 [Nov 9 18:20:31] a=rtpmap:0 PCMU/8000 [Nov 9 18:20:31] a=rtpmap:2 G726-32/8000 [Nov 9 18:20:31] a=rtpmap:3 GSM/8000 [Nov 9 18:20:31] a=rtpmap:97 iLBC/8000 [Nov 9 18:20:31] a=rtpmap:110 speex/8000 [Nov 9 18:20:31] a=rtpmap:111 speex/16000 [Nov 9 18:20:31] a=rtpmap:9 G722/8000 [Nov 9 18:20:31] a=rtpmap:101 telephone-event/8000 [Nov 9 18:20:31] a=fmtp:101 0-16 [Nov 9 18:20:31] a=ssrc:2516751377 [Nov 9 18:20:31] a=sendrecv [Nov 9 18:20:31] <-------------> [Nov 9 18:20:31] --- (16 headers 19 lines) --- [Nov 9 18:20:31] Sending to 192.168.210.111:5060 (no NAT) [Nov 9 18:20:31] Using INVITE request as basis request - 80D9017C-0EA5-E611-8500-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] Found peer '290' for '290' from 192.168.210.111:5060 [Nov 9 18:20:31] == Using SIP RTP CoS mark 5 [Nov 9 18:20:31] Found RTP audio format 107 [Nov 9 18:20:31] Found RTP audio format 8 [Nov 9 18:20:31] Found RTP audio format 0 [Nov 9 18:20:31] Found RTP audio format 2 [Nov 9 18:20:31] Found RTP audio format 3 [Nov 9 18:20:31] Found RTP audio format 97 [Nov 9 18:20:31] Found RTP audio format 110 [Nov 9 18:20:31] Found RTP audio format 111 [Nov 9 18:20:31] Found RTP audio format 9 [Nov 9 18:20:31] Found RTP audio format 101 [Nov 9 18:20:31] Found audio description format opus for ID 107 [Nov 9 18:20:31] Found audio description format PCMA for ID 8 [Nov 9 18:20:31] Found audio description format PCMU for ID 0 [Nov 9 18:20:31] Found audio description format G726-32 for ID 2 [Nov 9 18:20:31] Found audio description format GSM for ID 3 [Nov 9 18:20:31] Found audio description format iLBC for ID 97 [Nov 9 18:20:31] Found audio description format speex for ID 110 [Nov 9 18:20:31] Found audio description format speex for ID 111 [Nov 9 18:20:31] Found audio description format G722 for ID 9 [Nov 9 18:20:31] Found audio description format telephone-event for ID 101 [Nov 9 18:20:31] Capabilities: us - (alaw), peer - audio=(ulaw|gsm|alaw|g722|ilbc|opus|speex|speex16)/video=(nothing)/text=(nothing), combined - (alaw) [Nov 9 18:20:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 9 18:20:31] Peer audio RTP is at port 192.168.210.111:5064 [Nov 9 18:20:31] Peer doesn't provide T.140 [Nov 9 18:20:31] Looking for toronto101 in LocalSets (domain 192.168.210.132) [Nov 9 18:20:31] [Nov 9 18:20:31] <--- Reliably Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 9 18:20:31] SIP/2.0 404 Not Found [Nov 9 18:20:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80d9017c0ea5e61185025c199ea0d207;received=192.168.210.111;rport=5060 [Nov 9 18:20:31] From: "PhonerLite" ;tag=1111206135 [Nov 9 18:20:31] To: ;tag=as5ee80744 [Nov 9 18:20:31] Call-ID: 80D9017C-0EA5-E611-8500-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] CSeq: 139 INVITE [Nov 9 18:20:31] Server: Asterisk PBX GIT-master-0d85f18 [Nov 9 18:20:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 9 18:20:31] Supported: replaces, timer [Nov 9 18:20:31] Content-Length: 0 [Nov 9 18:20:31] [Nov 9 18:20:31] [Nov 9 18:20:31] <------------> [Nov 9 18:20:31] NOTICE[2240][C-0000000b]: chan_sip.c:26315 handle_request_invite: Call from '290' (192.168.210.111:5060) to extension 'toronto101' rejected because extension not found in context 'LocalSets'. [Nov 9 18:20:31] Scheduling destruction of SIP dialog '80D9017C-0EA5-E611-8500-5C199EA0D207@192.168.210.111' in 32000 ms (Method: INVITE) [Nov 9 18:20:31] [Nov 9 18:20:31] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 9 18:20:31] ACK sip:toronto101@192.168.210.132 SIP/2.0 [Nov 9 18:20:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80d9017c0ea5e61185025c199ea0d207;rport [Nov 9 18:20:31] From: "PhonerLite" ;tag=1111206135 [Nov 9 18:20:31] To: ;tag=as5ee80744 [Nov 9 18:20:31] Call-ID: 80D9017C-0EA5-E611-8500-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] CSeq: 139 ACK [Nov 9 18:20:31] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="4a9d526e", uri="sip:toronto101@192.168.210.132", response="813c7e002a55294dbf52bed8e938834f", algorithm=MD5 [Nov 9 18:20:31] Content-Length: 0 [Nov 9 18:20:31] [Nov 9 18:20:31] <-------------> [Nov 9 18:20:31] --- (8 headers 0 lines) --- [Nov 9 18:20:31] [Nov 9 18:20:31] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 9 18:20:31] ACK sip:toronto101@192.168.210.132 SIP/2.0 [Nov 9 18:20:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80d9017c0ea5e61185025c199ea0d207;rport [Nov 9 18:20:31] From: "PhonerLite" ;tag=1111206135 [Nov 9 18:20:31] To: ;tag=as5ee80744 [Nov 9 18:20:31] Call-ID: 80D9017C-0EA5-E611-8500-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] CSeq: 139 ACK [Nov 9 18:20:31] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="4a9d526e", uri="sip:toronto101@192.168.210.132", response="813c7e002a55294dbf52bed8e938834f", algorithm=MD5 [Nov 9 18:20:31] Content-Length: 0 [Nov 9 18:20:31] [Nov 9 18:20:31] <-------------> [Nov 9 18:20:31] --- (8 headers 0 lines) --- [Nov 9 18:20:31] [Nov 9 18:20:31] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 9 18:20:31] BYE sip:6100@192.168.210.132:5060 SIP/2.0 [Nov 9 18:20:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80d9017c0ea5e61185035c199ea0d207;rport [Nov 9 18:20:31] From: "PhonerLite" ;tag=859153178 [Nov 9 18:20:31] To: ;tag=as4740f019 [Nov 9 18:20:31] Call-ID: 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] CSeq: 140 BYE [Nov 9 18:20:31] Contact: [Nov 9 18:20:31] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="42ad9e32", uri="sip:6100@192.168.210.132:5060", response="9c3dd4a31c86bee3b4d00562eabf4c13", algorithm=MD5 [Nov 9 18:20:31] Max-Forwards: 70 [Nov 9 18:20:31] User-Agent: SIPPER for PhonerLite [Nov 9 18:20:31] Content-Length: 0 [Nov 9 18:20:31] [Nov 9 18:20:31] <-------------> [Nov 9 18:20:31] --- (11 headers 0 lines) --- [Nov 9 18:20:31] Sending to 192.168.210.111:5060 (no NAT) [Nov 9 18:20:31] Scheduling destruction of SIP dialog '00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111' in 32000 ms (Method: BYE) [Nov 9 18:20:31] [Nov 9 18:20:31] <--- Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 9 18:20:31] SIP/2.0 200 OK [Nov 9 18:20:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80d9017c0ea5e61185035c199ea0d207;received=192.168.210.111;rport=5060 [Nov 9 18:20:31] From: "PhonerLite" ;tag=859153178 [Nov 9 18:20:31] To: ;tag=as4740f019 [Nov 9 18:20:31] Call-ID: 00E90679-0EA5-E611-84FD-5C199EA0D207@192.168.210.111 [Nov 9 18:20:31] CSeq: 140 BYE [Nov 9 18:20:31] Server: Asterisk PBX GIT-master-0d85f18 [Nov 9 18:20:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 9 18:20:31] Supported: replaces, timer [Nov 9 18:20:31] Content-Length: 0 [Nov 9 18:20:31] [Nov 9 18:20:31] [Nov 9 18:20:31] <------------> [Nov 9 18:20:40] [Nov 9 18:20:40] <--- SIP read from UDP:192.168.210.41:5060 ---> [Nov 9 18:20:40] [Nov 9 18:20:40] [Nov 9 18:20:40] <------------->