kaworu*CLI> sip set debug on SIP Debugging enabled [Jan 16 17:41:59] Retransmitting #7 (no NAT) to 173.208.168.34:5070: [Jan 16 17:41:59] SIP/2.0 401 Unauthorized [Jan 16 17:41:59] Via: SIP/2.0/UDP 173.208.168.34:5070;branch=z9hG4bK-bce48f4927fd0834c36002e0cbc94cd7;received=173.208.168.34;rport=5070 [Jan 16 17:41:59] From: 2000;tag=18988e88 [Jan 16 17:41:59] To: 700441904891104;tag=as35cd4ca0 [Jan 16 17:41:59] Call-ID: bce48f4927fd0834c36002e0cbc94cd7 [Jan 16 17:41:59] CSeq: 1 INVITE [Jan 16 17:41:59] Server: Asterisk PBX 13.1.0 [Jan 16 17:41:59] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:41:59] Supported: replaces, timer [Jan 16 17:41:59] WWW-Authenticate: Digest algorithm=MD5, realm="kaworu.kunbox.net", nonce="3c54d5ea" [Jan 16 17:41:59] Content-Length: 0 [Jan 16 17:41:59] [Jan 16 17:41:59] [Jan 16 17:41:59] --- [Jan 16 17:42:01] Reliably Transmitting (no NAT) to 192.168.0.51:5060: [Jan 16 17:42:01] OPTIONS sip:76@192.168.0.51:5060 SIP/2.0 [Jan 16 17:42:01] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK4709e04d [Jan 16 17:42:01] Max-Forwards: 70 [Jan 16 17:42:01] From: "asterisk" ;tag=as029f3da4 [Jan 16 17:42:01] To: [Jan 16 17:42:01] Contact: [Jan 16 17:42:01] Call-ID: 2d93f1595989853771771e87161a6c20@192.168.0.20:5060 [Jan 16 17:42:01] CSeq: 102 OPTIONS [Jan 16 17:42:01] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:01] Date: Fri, 16 Jan 2015 16:42:01 GMT [Jan 16 17:42:01] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:01] Supported: replaces, timer [Jan 16 17:42:01] Content-Length: 0 [Jan 16 17:42:01] [Jan 16 17:42:01] [Jan 16 17:42:01] --- [Jan 16 17:42:01] [Jan 16 17:42:01] <--- SIP read from UDP:192.168.0.51:5060 ---> [Jan 16 17:42:01] SIP/2.0 200 OK [Jan 16 17:42:01] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK4709e04d [Jan 16 17:42:01] From: "asterisk" ;tag=as029f3da4 [Jan 16 17:42:01] To: ;tag=5bxit3mdwd [Jan 16 17:42:01] Call-ID: 2d93f1595989853771771e87161a6c20@192.168.0.20:5060 [Jan 16 17:42:01] CSeq: 102 OPTIONS [Jan 16 17:42:01] Contact: ;reg-id=1 [Jan 16 17:42:01] User-Agent: snom360/8.7.3.25 [Jan 16 17:42:01] Accept-Language: en [Jan 16 17:42:01] Accept: application/sdp [Jan 16 17:42:01] Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Jan 16 17:42:01] Allow-Events: talk, hold, refer, call-info [Jan 16 17:42:01] Supported: timer, 100rel, replaces, from-change [Jan 16 17:42:01] Content-Length: 0 [Jan 16 17:42:01] [Jan 16 17:42:01] <-------------> [Jan 16 17:42:01] --- (14 headers 0 lines) --- [Jan 16 17:42:01] Really destroying SIP dialog '2d93f1595989853771771e87161a6c20@192.168.0.20:5060' Method: OPTIONS [Jan 16 17:42:03] Retransmitting #8 (no NAT) to 173.208.168.34:5070: [Jan 16 17:42:03] SIP/2.0 401 Unauthorized [Jan 16 17:42:03] Via: SIP/2.0/UDP 173.208.168.34:5070;branch=z9hG4bK-bce48f4927fd0834c36002e0cbc94cd7;received=173.208.168.34;rport=5070 [Jan 16 17:42:03] From: 2000;tag=18988e88 [Jan 16 17:42:03] To: 700441904891104;tag=as35cd4ca0 [Jan 16 17:42:03] Call-ID: bce48f4927fd0834c36002e0cbc94cd7 [Jan 16 17:42:03] CSeq: 1 INVITE [Jan 16 17:42:03] Server: Asterisk PBX 13.1.0 [Jan 16 17:42:03] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:03] Supported: replaces, timer [Jan 16 17:42:03] WWW-Authenticate: Digest algorithm=MD5, realm="kaworu.kunbox.net", nonce="3c54d5ea" [Jan 16 17:42:03] Content-Length: 0 [Jan 16 17:42:03] [Jan 16 17:42:03] [Jan 16 17:42:03] --- [Jan 16 17:42:04] Reliably Transmitting (no NAT) to 192.168.0.180:62214: [Jan 16 17:42:04] OPTIONS sip:handy-felix@192.168.0.180:62214;app-id=org.linphone.phone.prod;pn-type=apple;pn-tok=1A4629A367797941D497C8662E11391B2C79C574B4698F6C723A72CBD140A4FF;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf SIP/2.0 [Jan 16 17:42:04] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK5d4865f0 [Jan 16 17:42:04] Max-Forwards: 70 [Jan 16 17:42:04] From: "asterisk" ;tag=as0d012382 [Jan 16 17:42:04] To: [Jan 16 17:42:04] Contact: [Jan 16 17:42:04] Call-ID: 4189a6094a7c7c0e7a13364f57185ecd@192.168.0.20:5060 [Jan 16 17:42:04] CSeq: 102 OPTIONS [Jan 16 17:42:04] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:04] Date: Fri, 16 Jan 2015 16:42:04 GMT [Jan 16 17:42:04] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:04] Supported: replaces, timer [Jan 16 17:42:04] Content-Length: 0 [Jan 16 17:42:04] [Jan 16 17:42:04] [Jan 16 17:42:04] --- [Jan 16 17:42:05] Retransmitting #1 (no NAT) to 192.168.0.180:62214: [Jan 16 17:42:05] OPTIONS sip:handy-felix@192.168.0.180:62214;app-id=org.linphone.phone.prod;pn-type=apple;pn-tok=1A4629A367797941D497C8662E11391B2C79C574B4698F6C723A72CBD140A4FF;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf SIP/2.0 [Jan 16 17:42:05] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK5d4865f0 [Jan 16 17:42:05] Max-Forwards: 70 [Jan 16 17:42:05] From: "asterisk" ;tag=as0d012382 [Jan 16 17:42:05] To: [Jan 16 17:42:05] Contact: [Jan 16 17:42:05] Call-ID: 4189a6094a7c7c0e7a13364f57185ecd@192.168.0.20:5060 [Jan 16 17:42:05] CSeq: 102 OPTIONS [Jan 16 17:42:05] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:05] Date: Fri, 16 Jan 2015 16:42:04 GMT [Jan 16 17:42:05] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:05] Supported: replaces, timer [Jan 16 17:42:05] Content-Length: 0 [Jan 16 17:42:05] [Jan 16 17:42:05] [Jan 16 17:42:05] --- [Jan 16 17:42:06] Retransmitting #2 (no NAT) to 192.168.0.180:62214: [Jan 16 17:42:06] OPTIONS sip:handy-felix@192.168.0.180:62214;app-id=org.linphone.phone.prod;pn-type=apple;pn-tok=1A4629A367797941D497C8662E11391B2C79C574B4698F6C723A72CBD140A4FF;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf SIP/2.0 [Jan 16 17:42:06] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK5d4865f0 [Jan 16 17:42:06] Max-Forwards: 70 [Jan 16 17:42:06] From: "asterisk" ;tag=as0d012382 [Jan 16 17:42:06] To: [Jan 16 17:42:06] Contact: [Jan 16 17:42:06] Call-ID: 4189a6094a7c7c0e7a13364f57185ecd@192.168.0.20:5060 [Jan 16 17:42:06] CSeq: 102 OPTIONS [Jan 16 17:42:06] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:06] Date: Fri, 16 Jan 2015 16:42:04 GMT [Jan 16 17:42:06] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:06] Supported: replaces, timer [Jan 16 17:42:06] Content-Length: 0 [Jan 16 17:42:06] [Jan 16 17:42:06] [Jan 16 17:42:06] --- [Jan 16 17:42:06] [Jan 16 17:42:06] <--- SIP read from UDP:192.168.0.51:5060 ---> [Jan 16 17:42:06] INVITE sip:asterisk@192.168.0.20 SIP/2.0 [Jan 16 17:42:06] Via: SIP/2.0/UDP 192.168.0.51:5060;branch=z9hG4bK-6bd9y826wkz3;rport [Jan 16 17:42:06] From: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:06] To: [Jan 16 17:42:06] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:06] CSeq: 1 INVITE [Jan 16 17:42:06] Max-Forwards: 70 [Jan 16 17:42:06] Contact: ;reg-id=1 [Jan 16 17:42:06] X-Serialnumber: 00041329C09F [Jan 16 17:42:06] P-Key-Flags: resolution="31x13", keys="4" [Jan 16 17:42:06] User-Agent: snom360/8.7.3.25 [Jan 16 17:42:06] Accept: application/sdp [Jan 16 17:42:06] Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Jan 16 17:42:06] Allow-Events: talk, hold, refer, call-info [Jan 16 17:42:06] Supported: timer, 100rel, replaces, from-change [Jan 16 17:42:06] Session-Expires: 3600;refresher=uas [Jan 16 17:42:06] Min-SE: 90 [Jan 16 17:42:06] Content-Type: application/sdp [Jan 16 17:42:06] Content-Length: 485 [Jan 16 17:42:06] [Jan 16 17:42:06] v=0 [Jan 16 17:42:06] o=root 877302269 877302269 IN IP4 192.168.0.51 [Jan 16 17:42:06] s=call [Jan 16 17:42:06] c=IN IP4 192.168.0.51 [Jan 16 17:42:06] t=0 0 [Jan 16 17:42:06] m=audio 54500 RTP/AVP 9 0 8 3 99 108 18 101 [Jan 16 17:42:06] a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:RDhfxyHA0+MJJBphr4/SoWz4nkGf7t3nLMZm7Kdk [Jan 16 17:42:06] a=rtpmap:9 G722/8000 [Jan 16 17:42:06] a=rtpmap:0 PCMU/8000 [Jan 16 17:42:06] a=rtpmap:8 PCMA/8000 [Jan 16 17:42:06] a=rtpmap:3 GSM/8000 [Jan 16 17:42:06] a=rtpmap:99 G726-32/8000 [Jan 16 17:42:06] a=rtpmap:108 AAL2-G726-32/8000 [Jan 16 17:42:06] a=rtpmap:18 G729/8000 [Jan 16 17:42:06] a=fmtp:18 annexb=no [Jan 16 17:42:06] a=rtpmap:101 telephone-event/8000 [Jan 16 17:42:06] a=fmtp:101 0-15 [Jan 16 17:42:06] a=ptime:20 [Jan 16 17:42:06] a=sendrecv [Jan 16 17:42:06] <-------------> [Jan 16 17:42:06] --- (19 headers 19 lines) --- [Jan 16 17:42:06] Sending to 192.168.0.51:5060 (no NAT) [Jan 16 17:42:06] Sending to 192.168.0.51:5060 (no NAT) [Jan 16 17:42:06] Using INVITE request as basis request - 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:06] Found peer '76' for '76' from 192.168.0.51:5060 [Jan 16 17:42:06] [Jan 16 17:42:06] <--- Reliably Transmitting (no NAT) to 192.168.0.51:5060 ---> [Jan 16 17:42:06] SIP/2.0 401 Unauthorized [Jan 16 17:42:06] Via: SIP/2.0/UDP 192.168.0.51:5060;branch=z9hG4bK-6bd9y826wkz3;received=192.168.0.51;rport=5060 [Jan 16 17:42:06] From: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:06] To: ;tag=as5df06dbe [Jan 16 17:42:06] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:06] CSeq: 1 INVITE [Jan 16 17:42:06] Server: Asterisk PBX 13.1.0 [Jan 16 17:42:06] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:06] Supported: replaces, timer [Jan 16 17:42:06] WWW-Authenticate: Digest algorithm=MD5, realm="kaworu.kunbox.net", nonce="53cb07ad" [Jan 16 17:42:06] Content-Length: 0 [Jan 16 17:42:06] [Jan 16 17:42:06] [Jan 16 17:42:06] <------------> [Jan 16 17:42:06] Scheduling destruction of SIP dialog '54b93f5e0952-1zsswll6gkyr' in 6400 ms (Method: INVITE) [Jan 16 17:42:06] [Jan 16 17:42:06] <--- SIP read from UDP:192.168.0.51:5060 ---> [Jan 16 17:42:06] ACK sip:asterisk@192.168.0.20 SIP/2.0 [Jan 16 17:42:06] Via: SIP/2.0/UDP 192.168.0.51:5060;branch=z9hG4bK-6bd9y826wkz3;rport [Jan 16 17:42:06] From: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:06] To: ;tag=as5df06dbe [Jan 16 17:42:06] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:06] CSeq: 1 ACK [Jan 16 17:42:06] Max-Forwards: 70 [Jan 16 17:42:06] Contact: ;reg-id=1 [Jan 16 17:42:06] Content-Length: 0 [Jan 16 17:42:06] [Jan 16 17:42:06] <-------------> [Jan 16 17:42:06] --- (9 headers 0 lines) --- [Jan 16 17:42:07] Reliably Transmitting (no NAT) to 192.168.0.50:5061: [Jan 16 17:42:07] OPTIONS sip:spa3102-pstn@192.168.0.50:5061 SIP/2.0 [Jan 16 17:42:07] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK46ae0331 [Jan 16 17:42:07] Max-Forwards: 70 [Jan 16 17:42:07] From: "asterisk" ;tag=as553c1f1a [Jan 16 17:42:07] To: [Jan 16 17:42:07] Contact: [Jan 16 17:42:07] Call-ID: 3a0a175e023a20284563641174710b53@192.168.0.20:5060 [Jan 16 17:42:07] CSeq: 102 OPTIONS [Jan 16 17:42:07] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:07] Date: Fri, 16 Jan 2015 16:42:07 GMT [Jan 16 17:42:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:07] Supported: replaces, timer [Jan 16 17:42:07] Content-Length: 0 [Jan 16 17:42:07] [Jan 16 17:42:07] [Jan 16 17:42:07] --- [Jan 16 17:42:07] Retransmitting #3 (no NAT) to 192.168.0.180:62214: [Jan 16 17:42:07] OPTIONS sip:handy-felix@192.168.0.180:62214;app-id=org.linphone.phone.prod;pn-type=apple;pn-tok=1A4629A367797941D497C8662E11391B2C79C574B4698F6C723A72CBD140A4FF;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf SIP/2.0 [Jan 16 17:42:07] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK5d4865f0 [Jan 16 17:42:07] Max-Forwards: 70 [Jan 16 17:42:07] From: "asterisk" ;tag=as0d012382 [Jan 16 17:42:07] To: [Jan 16 17:42:07] Contact: [Jan 16 17:42:07] Call-ID: 4189a6094a7c7c0e7a13364f57185ecd@192.168.0.20:5060 [Jan 16 17:42:07] CSeq: 102 OPTIONS [Jan 16 17:42:07] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:07] Date: Fri, 16 Jan 2015 16:42:04 GMT [Jan 16 17:42:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:07] Supported: replaces, timer [Jan 16 17:42:07] Content-Length: 0 [Jan 16 17:42:07] [Jan 16 17:42:07] [Jan 16 17:42:07] --- [Jan 16 17:42:07] Retransmitting #9 (no NAT) to 173.208.168.34:5070: [Jan 16 17:42:07] SIP/2.0 401 Unauthorized [Jan 16 17:42:07] Via: SIP/2.0/UDP 173.208.168.34:5070;branch=z9hG4bK-bce48f4927fd0834c36002e0cbc94cd7;received=173.208.168.34;rport=5070 [Jan 16 17:42:07] From: 2000;tag=18988e88 [Jan 16 17:42:07] To: 700441904891104;tag=as35cd4ca0 [Jan 16 17:42:07] Call-ID: bce48f4927fd0834c36002e0cbc94cd7 [Jan 16 17:42:07] CSeq: 1 INVITE [Jan 16 17:42:07] Server: Asterisk PBX 13.1.0 [Jan 16 17:42:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:07] Supported: replaces, timer [Jan 16 17:42:07] WWW-Authenticate: Digest algorithm=MD5, realm="kaworu.kunbox.net", nonce="3c54d5ea" [Jan 16 17:42:07] Content-Length: 0 [Jan 16 17:42:07] [Jan 16 17:42:07] [Jan 16 17:42:07] --- [Jan 16 17:42:08] Retransmitting #1 (no NAT) to 192.168.0.50:5061: [Jan 16 17:42:08] OPTIONS sip:spa3102-pstn@192.168.0.50:5061 SIP/2.0 [Jan 16 17:42:08] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK46ae0331 [Jan 16 17:42:08] Max-Forwards: 70 [Jan 16 17:42:08] From: "asterisk" ;tag=as553c1f1a [Jan 16 17:42:08] To: [Jan 16 17:42:08] Contact: [Jan 16 17:42:08] Call-ID: 3a0a175e023a20284563641174710b53@192.168.0.20:5060 [Jan 16 17:42:08] CSeq: 102 OPTIONS [Jan 16 17:42:08] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:08] Date: Fri, 16 Jan 2015 16:42:07 GMT [Jan 16 17:42:08] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:08] Supported: replaces, timer [Jan 16 17:42:08] Content-Length: 0 [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] --- [Jan 16 17:42:08] [Jan 16 17:42:08] <--- SIP read from UDP:192.168.0.51:5060 ---> [Jan 16 17:42:08] INVITE sip:asterisk@192.168.0.20 SIP/2.0 [Jan 16 17:42:08] Via: SIP/2.0/UDP 192.168.0.51:5060;branch=z9hG4bK-5rbvl76dfyep;rport [Jan 16 17:42:08] From: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:08] To: [Jan 16 17:42:08] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:08] CSeq: 2 INVITE [Jan 16 17:42:08] Max-Forwards: 70 [Jan 16 17:42:08] Contact: ;reg-id=1 [Jan 16 17:42:08] X-Serialnumber: 00041329C09F [Jan 16 17:42:08] P-Key-Flags: resolution="31x13", keys="4" [Jan 16 17:42:08] User-Agent: snom360/8.7.3.25 [Jan 16 17:42:08] Accept: application/sdp [Jan 16 17:42:08] Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE [Jan 16 17:42:08] Allow-Events: talk, hold, refer, call-info [Jan 16 17:42:08] Supported: timer, 100rel, replaces, from-change [Jan 16 17:42:08] Session-Expires: 3600;refresher=uas [Jan 16 17:42:08] Min-SE: 90 [Jan 16 17:42:08] Authorization: Digest username="76",realm="kaworu.kunbox.net",nonce="53cb07ad",uri="sip:asterisk@192.168.0.20",response="bfc2c88a34bb57cd392529fd490567c9",algorithm=MD5 [Jan 16 17:42:08] Content-Type: application/sdp [Jan 16 17:42:08] Content-Length: 485 [Jan 16 17:42:08] [Jan 16 17:42:08] v=0 [Jan 16 17:42:08] o=root 877302269 877302269 IN IP4 192.168.0.51 [Jan 16 17:42:08] s=call [Jan 16 17:42:08] c=IN IP4 192.168.0.51 [Jan 16 17:42:08] t=0 0 [Jan 16 17:42:08] m=audio 54500 RTP/AVP 9 0 8 3 99 108 18 101 [Jan 16 17:42:08] a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:RDhfxyHA0+MJJBphr4/SoWz4nkGf7t3nLMZm7Kdk [Jan 16 17:42:08] a=rtpmap:9 G722/8000 [Jan 16 17:42:08] a=rtpmap:0 PCMU/8000 [Jan 16 17:42:08] a=rtpmap:8 PCMA/8000 [Jan 16 17:42:08] a=rtpmap:3 GSM/8000 [Jan 16 17:42:08] a=rtpmap:99 G726-32/8000 [Jan 16 17:42:08] a=rtpmap:108 AAL2-G726-32/8000 [Jan 16 17:42:08] a=rtpmap:18 G729/8000 [Jan 16 17:42:08] a=fmtp:18 annexb=no [Jan 16 17:42:08] a=rtpmap:101 telephone-event/8000 [Jan 16 17:42:08] a=fmtp:101 0-15 [Jan 16 17:42:08] a=ptime:20 [Jan 16 17:42:08] a=sendrecv [Jan 16 17:42:08] <-------------> [Jan 16 17:42:08] --- (20 headers 19 lines) --- [Jan 16 17:42:08] Sending to 192.168.0.51:5060 (no NAT) [Jan 16 17:42:08] Using INVITE request as basis request - 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:08] Found peer '76' for '76' from 192.168.0.51:5060 [Jan 16 17:42:08] == Using SIP RTP CoS mark 5 [Jan 16 17:42:08] Found RTP audio format 9 [Jan 16 17:42:08] Found RTP audio format 0 [Jan 16 17:42:08] Found RTP audio format 8 [Jan 16 17:42:08] Found RTP audio format 3 [Jan 16 17:42:08] Found RTP audio format 99 [Jan 16 17:42:08] Found RTP audio format 108 [Jan 16 17:42:08] Found RTP audio format 18 [Jan 16 17:42:08] Found RTP audio format 101 [Jan 16 17:42:08] Found audio description format G722 for ID 9 [Jan 16 17:42:08] Found audio description format PCMU for ID 0 [Jan 16 17:42:08] Found audio description format PCMA for ID 8 [Jan 16 17:42:08] Found audio description format GSM for ID 3 [Jan 16 17:42:08] Found audio description format G726-32 for ID 99 [Jan 16 17:42:08] Found audio description format AAL2-G726-32 for ID 108 [Jan 16 17:42:08] Found audio description format G729 for ID 18 [Jan 16 17:42:08] Found audio description format telephone-event for ID 101 [Jan 16 17:42:08] failed to extend from 64 to 98 [Jan 16 17:42:08] Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw|gsm|alaw|g722|g729|g726|g726aal2)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g726|g726aal2|g729|g722) [Jan 16 17:42:08] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 16 17:42:08] Peer audio RTP is at port 192.168.0.51:54500 [Jan 16 17:42:08] Looking for asterisk in internal (domain 192.168.0.20) [Jan 16 17:42:08] set_destination: Parsing for address/port to send to [Jan 16 17:42:08] set_destination: set destination to 192.168.0.52:5060 [Jan 16 17:42:08] Reliably Transmitting (no NAT) to 192.168.0.52:5060: [Jan 16 17:42:08] NOTIFY sip:91@192.168.0.52:5060 SIP/2.0 [Jan 16 17:42:08] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK13903edd;rport [Jan 16 17:42:08] Max-Forwards: 70 [Jan 16 17:42:08] From: ;tag=as1254995d [Jan 16 17:42:08] To: ;tag=5vsgks96rs [Jan 16 17:42:08] Contact: [Jan 16 17:42:08] Call-ID: 54b935d06aab-7wxqk99jossv [Jan 16 17:42:08] CSeq: 124 NOTIFY [Jan 16 17:42:08] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:08] Subscription-State: active [Jan 16 17:42:08] Event: dialog [Jan 16 17:42:08] Content-Type: application/dialog-info+xml [Jan 16 17:42:08] Content-Length: 201 [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] confirmed [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] --- [Jan 16 17:42:08] == Extension Changed 76[SoftPhone] new state InUse for Notify User 91 [Jan 16 17:42:08] set_destination: Parsing for address/port to send to [Jan 16 17:42:08] set_destination: set destination to 192.168.0.51:5060 [Jan 16 17:42:08] Reliably Transmitting (no NAT) to 192.168.0.51:5060: [Jan 16 17:42:08] NOTIFY sip:76@192.168.0.51:5060 SIP/2.0 [Jan 16 17:42:08] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK6d032626;rport [Jan 16 17:42:08] Max-Forwards: 70 [Jan 16 17:42:08] From: ;tag=as02d09b91 [Jan 16 17:42:08] To: ;tag=sytb991fg3 [Jan 16 17:42:08] Contact: [Jan 16 17:42:08] Call-ID: 54b938661fd9-rg4njam42org [Jan 16 17:42:08] CSeq: 120 NOTIFY [Jan 16 17:42:08] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:08] Subscription-State: active [Jan 16 17:42:08] Event: dialog [Jan 16 17:42:08] Content-Type: application/dialog-info+xml [Jan 16 17:42:08] Content-Length: 201 [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] confirmed [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] --- [Jan 16 17:42:08] == Extension Changed 76[SoftPhone] new state InUse for Notify User 76 [Jan 16 17:42:08] sip_route_dump: route/path hop: [Jan 16 17:42:08] [Jan 16 17:42:08] <--- Transmitting (no NAT) to 192.168.0.51:5060 ---> [Jan 16 17:42:08] SIP/2.0 100 Trying [Jan 16 17:42:08] Via: SIP/2.0/UDP 192.168.0.51:5060;branch=z9hG4bK-5rbvl76dfyep;received=192.168.0.51;rport=5060 [Jan 16 17:42:08] From: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:08] To: [Jan 16 17:42:08] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:08] CSeq: 2 INVITE [Jan 16 17:42:08] Server: Asterisk PBX 13.1.0 [Jan 16 17:42:08] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:08] Supported: replaces, timer [Jan 16 17:42:08] Session-Expires: 1800;refresher=uas [Jan 16 17:42:08] Contact: [Jan 16 17:42:08] Content-Length: 0 [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] <------------> [Jan 16 17:42:08] -- Executing [asterisk@internal:1] Goto("SIP/76-0000006f", "SoftPhone,98,1") in new stack [Jan 16 17:42:08] -- Goto (SoftPhone,98,1) [Jan 16 17:42:08] -- Executing [98@SoftPhone:1] VoiceMailMain("SIP/76-0000006f", "800,s") in new stack [Jan 16 17:42:08] Audio is at 7600 [Jan 16 17:42:08] Adding codec ulaw to SDP [Jan 16 17:42:08] Adding codec alaw to SDP [Jan 16 17:42:08] Adding codec gsm to SDP [Jan 16 17:42:08] Adding codec g726 to SDP [Jan 16 17:42:08] Adding codec g726aal2 to SDP [Jan 16 17:42:08] Adding codec g729 to SDP [Jan 16 17:42:08] Adding codec g722 to SDP [Jan 16 17:42:08] Adding codec g723 to SDP [Jan 16 17:42:08] Adding codec adpcm to SDP [Jan 16 17:42:08] Adding codec slin to SDP [Jan 16 17:42:08] Adding codec slin to SDP [Jan 16 17:42:08] Adding codec slin to SDP [Jan 16 17:42:08] Adding codec slin to SDP [Jan 16 17:42:08] Adding codec slin to SDP [Jan 16 17:42:08] Adding codec slin to SDP [Jan 16 17:42:08] Adding codec slin to SDP [Jan 16 17:42:08] Adding codec slin to SDP [Jan 16 17:42:08] Adding codec slin to SDP [Jan 16 17:42:08] Adding codec lpc10 to SDP [Jan 16 17:42:08] Adding codec speex to SDP [Jan 16 17:42:08] Adding codec speex to SDP [Jan 16 17:42:08] Adding codec speex to SDP [Jan 16 17:42:08] Adding codec ilbc to SDP [Jan 16 17:42:08] Adding codec siren7 to SDP [Jan 16 17:42:08] Adding codec siren14 to SDP [Jan 16 17:42:08] Adding codec testlaw to SDP [Jan 16 17:42:08] Adding codec g719 to SDP [Jan 16 17:42:08] Adding codec opus to SDP [Jan 16 17:42:08] Adding codec none to SDP [Jan 16 17:42:08] Adding non-codec 0x1 (telephone-event) to SDP [Jan 16 17:42:08] [Jan 16 17:42:08] <--- Reliably Transmitting (no NAT) to 192.168.0.51:5060 ---> [Jan 16 17:42:08] SIP/2.0 200 OK [Jan 16 17:42:08] Via: SIP/2.0/UDP 192.168.0.51:5060;branch=z9hG4bK-5rbvl76dfyep;received=192.168.0.51;rport=5060 [Jan 16 17:42:08] From: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:08] To: ;tag=as4e86b1b1 [Jan 16 17:42:08] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:08] CSeq: 2 INVITE [Jan 16 17:42:08] Server: Asterisk PBX 13.1.0 [Jan 16 17:42:08] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:08] Supported: replaces, timer [Jan 16 17:42:08] Session-Expires: 1800;refresher=uas [Jan 16 17:42:08] Contact: [Jan 16 17:42:08] Content-Type: application/sdp [Jan 16 17:42:08] Require: timer [Jan 16 17:42:08] Content-Length: 895 [Jan 16 17:42:08] [Jan 16 17:42:08] v=0 [Jan 16 17:42:08] o=root 1682995340 1682995340 IN IP4 192.168.0.20 [Jan 16 17:42:08] s=Asterisk PBX 13.1.0 [Jan 16 17:42:08] c=IN IP4 192.168.0.20 [Jan 16 17:42:08] t=0 0 [Jan 16 17:42:08] m=audio 7600 RTP/AVP 0 8 3 99 108 18 9 4 5 10 118 7 110 117 119 97 102 115 116 107 101 [Jan 16 17:42:08] a=rtpmap:0 PCMU/8000 [Jan 16 17:42:08] a=rtpmap:8 PCMA/8000 [Jan 16 17:42:08] a=rtpmap:3 GSM/8000 [Jan 16 17:42:08] a=rtpmap:99 G726-32/8000 [Jan 16 17:42:08] a=rtpmap:108 AAL2-G726-32/8000 [Jan 16 17:42:08] a=rtpmap:18 G729/8000 [Jan 16 17:42:08] a=fmtp:18 annexb=no [Jan 16 17:42:08] a=rtpmap:9 G722/8000 [Jan 16 17:42:08] a=rtpmap:4 G723/8000 [Jan 16 17:42:08] a=fmtp:4 annexa=no [Jan 16 17:42:08] a=rtpmap:5 DVI4/8000 [Jan 16 17:42:08] a=rtpmap:10 L16/8000 [Jan 16 17:42:08] a=rtpmap:118 L16/16000 [Jan 16 17:42:08] a=rtpmap:7 LPC/8000 [Jan 16 17:42:08] a=rtpmap:110 speex/8000 [Jan 16 17:42:08] a=rtpmap:117 speex/16000 [Jan 16 17:42:08] a=rtpmap:119 speex/32000 [Jan 16 17:42:08] a=rtpmap:97 iLBC/8000 [Jan 16 17:42:08] a=fmtp:97 mode=0 [Jan 16 17:42:08] a=rtpmap:102 G7221/16000 [Jan 16 17:42:08] a=fmtp:102 bitrate=32000 [Jan 16 17:42:08] a=rtpmap:115 G7221/32000 [Jan 16 17:42:08] a=fmtp:115 bitrate=48000 [Jan 16 17:42:08] a=rtpmap:116 G719/48000 [Jan 16 17:42:08] a=fmtp:116 bitrate=64000 [Jan 16 17:42:08] a=rtpmap:107 opus/48000/2 [Jan 16 17:42:08] a=rtpmap:101 telephone-event/8000 [Jan 16 17:42:08] a=fmtp:101 0-16 [Jan 16 17:42:08] a=maxptime:20 [Jan 16 17:42:08] a=sendrecv [Jan 16 17:42:08] [Jan 16 17:42:08] <------------> [Jan 16 17:42:08] [Jan 16 17:42:08] <--- SIP read from UDP:192.168.0.52:5060 ---> [Jan 16 17:42:08] SIP/2.0 200 Ok [Jan 16 17:42:08] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK13903edd;rport=5060 [Jan 16 17:42:08] From: ;tag=as1254995d [Jan 16 17:42:08] To: ;tag=5vsgks96rs [Jan 16 17:42:08] Call-ID: 54b935d06aab-7wxqk99jossv [Jan 16 17:42:08] CSeq: 124 NOTIFY [Jan 16 17:42:08] Content-Length: 0 [Jan 16 17:42:08] [Jan 16 17:42:08] <-------------> [Jan 16 17:42:08] --- (7 headers 0 lines) --- [Jan 16 17:42:08] Retransmitting #1 (no NAT) to 192.168.0.51:5060: [Jan 16 17:42:08] SIP/2.0 200 OK [Jan 16 17:42:08] Via: SIP/2.0/UDP 192.168.0.51:5060;branch=z9hG4bK-5rbvl76dfyep;received=192.168.0.51;rport=5060 [Jan 16 17:42:08] From: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:08] To: ;tag=as4e86b1b1 [Jan 16 17:42:08] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:08] CSeq: 2 INVITE [Jan 16 17:42:08] Server: Asterisk PBX 13.1.0 [Jan 16 17:42:08] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:08] Supported: replaces, timer [Jan 16 17:42:08] Session-Expires: 1800;refresher=uas [Jan 16 17:42:08] Contact: [Jan 16 17:42:08] Content-Type: application/sdp [Jan 16 17:42:08] Require: timer [Jan 16 17:42:08] Content-Length: 895 [Jan 16 17:42:08] [Jan 16 17:42:08] v=0 [Jan 16 17:42:08] o=root 1682995340 1682995340 IN IP4 192.168.0.20 [Jan 16 17:42:08] s=Asterisk PBX 13.1.0 [Jan 16 17:42:08] c=IN IP4 192.168.0.20 [Jan 16 17:42:08] t=0 0 [Jan 16 17:42:08] m=audio 7600 RTP/AVP 0 8 3 99 108 18 9 4 5 10 118 7 110 117 119 97 102 115 116 107 101 [Jan 16 17:42:08] a=rtpmap:0 PCMU/8000 [Jan 16 17:42:08] a=rtpmap:8 PCMA/8000 [Jan 16 17:42:08] a=rtpmap:3 GSM/8000 [Jan 16 17:42:08] a=rtpmap:99 G726-32/8000 [Jan 16 17:42:08] a=rtpmap:108 AAL2-G726-32/8000 [Jan 16 17:42:08] a=rtpmap:18 G729/8000 [Jan 16 17:42:08] a=fmtp:18 annexb=no [Jan 16 17:42:08] a=rtpmap:9 G722/8000 [Jan 16 17:42:08] a=rtpmap:4 G723/8000 [Jan 16 17:42:08] a=fmtp:4 annexa=no [Jan 16 17:42:08] a=rtpmap:5 DVI4/8000 [Jan 16 17:42:08] a=rtpmap:10 L16/8000 [Jan 16 17:42:08] a=rtpmap:118 L16/16000 [Jan 16 17:42:08] a=rtpmap:7 LPC/8000 [Jan 16 17:42:08] a=rtpmap:110 speex/8000 [Jan 16 17:42:08] a=rtpmap:117 speex/16000 [Jan 16 17:42:08] a=rtpmap:119 speex/32000 [Jan 16 17:42:08] a=rtpmap:97 iLBC/8000 [Jan 16 17:42:08] a=fmtp:97 mode=0 [Jan 16 17:42:08] a=rtpmap:102 G7221/16000 [Jan 16 17:42:08] a=fmtp:102 bitrate=32000 [Jan 16 17:42:08] a=rtpmap:115 G7221/32000 [Jan 16 17:42:08] a=fmtp:115 bitrate=48000 [Jan 16 17:42:08] a=rtpmap:116 G719/48000 [Jan 16 17:42:08] a=fmtp:116 bitrate=64000 [Jan 16 17:42:08] a=rtpmap:107 opus/48000/2 [Jan 16 17:42:08] a=rtpmap:101 telephone-event/8000 [Jan 16 17:42:08] a=fmtp:101 0-16 [Jan 16 17:42:08] a=maxptime:20 [Jan 16 17:42:08] a=sendrecv [Jan 16 17:42:08] [Jan 16 17:42:08] --- [Jan 16 17:42:08] [Jan 16 17:42:08] <--- SIP read from UDP:192.168.0.51:5060 ---> [Jan 16 17:42:08] ACK sip:asterisk@192.168.0.20:5060 SIP/2.0 [Jan 16 17:42:08] Via: SIP/2.0/UDP 192.168.0.51:5060;branch=z9hG4bK-98c79ky4y7mq;rport [Jan 16 17:42:08] From: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:08] To: ;tag=as4e86b1b1 [Jan 16 17:42:08] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:08] CSeq: 2 ACK [Jan 16 17:42:08] Max-Forwards: 70 [Jan 16 17:42:08] Contact: ;reg-id=1 [Jan 16 17:42:08] Content-Length: 0 [Jan 16 17:42:08] [Jan 16 17:42:08] <-------------> [Jan 16 17:42:08] --- (9 headers 0 lines) --- [Jan 16 17:42:08] Retransmitting #4 (no NAT) to 192.168.0.180:62214: [Jan 16 17:42:08] OPTIONS sip:handy-felix@192.168.0.180:62214;app-id=org.linphone.phone.prod;pn-type=apple;pn-tok=1A4629A367797941D497C8662E11391B2C79C574B4698F6C723A72CBD140A4FF;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf SIP/2.0 [Jan 16 17:42:08] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK5d4865f0 [Jan 16 17:42:08] Max-Forwards: 70 [Jan 16 17:42:08] From: "asterisk" ;tag=as0d012382 [Jan 16 17:42:08] To: [Jan 16 17:42:08] Contact: [Jan 16 17:42:08] Call-ID: 4189a6094a7c7c0e7a13364f57185ecd@192.168.0.20:5060 [Jan 16 17:42:08] CSeq: 102 OPTIONS [Jan 16 17:42:08] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:08] Date: Fri, 16 Jan 2015 16:42:04 GMT [Jan 16 17:42:08] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:08] Supported: replaces, timer [Jan 16 17:42:08] Content-Length: 0 [Jan 16 17:42:08] [Jan 16 17:42:08] [Jan 16 17:42:08] --- [Jan 16 17:42:08] Really destroying SIP dialog '4189a6094a7c7c0e7a13364f57185ecd@192.168.0.20:5060' Method: OPTIONS [Jan 16 17:42:08] > 0x7f888055a9a0 -- Probation passed - setting RTP source address to 192.168.0.51:54500 [Jan 16 17:42:08] -- Playing 'vm-youhave.ulaw' (language 'en') [Jan 16 17:42:08] [Jan 16 17:42:08] <--- SIP read from UDP:192.168.0.51:5060 ---> [Jan 16 17:42:08] ACK sip:asterisk@192.168.0.20:5060 SIP/2.0 [Jan 16 17:42:08] Via: SIP/2.0/UDP 192.168.0.51:5060;branch=z9hG4bK-98c79ky4y7mq;rport [Jan 16 17:42:08] From: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:08] To: ;tag=as4e86b1b1 [Jan 16 17:42:08] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:08] CSeq: 2 ACK [Jan 16 17:42:08] Max-Forwards: 70 [Jan 16 17:42:08] Contact: ;reg-id=1 [Jan 16 17:42:08] Content-Length: 0 [Jan 16 17:42:08] [Jan 16 17:42:08] <-------------> [Jan 16 17:42:08] --- (9 headers 0 lines) --- [Jan 16 17:42:09] Retransmitting #2 (no NAT) to 192.168.0.50:5061: [Jan 16 17:42:09] OPTIONS sip:spa3102-pstn@192.168.0.50:5061 SIP/2.0 [Jan 16 17:42:09] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK46ae0331 [Jan 16 17:42:09] Max-Forwards: 70 [Jan 16 17:42:09] From: "asterisk" ;tag=as553c1f1a [Jan 16 17:42:09] To: [Jan 16 17:42:09] Contact: [Jan 16 17:42:09] Call-ID: 3a0a175e023a20284563641174710b53@192.168.0.20:5060 [Jan 16 17:42:09] CSeq: 102 OPTIONS [Jan 16 17:42:09] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:09] Date: Fri, 16 Jan 2015 16:42:07 GMT [Jan 16 17:42:09] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:09] Supported: replaces, timer [Jan 16 17:42:09] Content-Length: 0 [Jan 16 17:42:09] [Jan 16 17:42:09] [Jan 16 17:42:09] --- [Jan 16 17:42:09] Retransmitting #1 (no NAT) to 192.168.0.51:5060: [Jan 16 17:42:09] NOTIFY sip:76@192.168.0.51:5060 SIP/2.0 [Jan 16 17:42:09] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK6d032626;rport [Jan 16 17:42:09] Max-Forwards: 70 [Jan 16 17:42:09] From: ;tag=as02d09b91 [Jan 16 17:42:09] To: ;tag=sytb991fg3 [Jan 16 17:42:09] Contact: [Jan 16 17:42:09] Call-ID: 54b938661fd9-rg4njam42org [Jan 16 17:42:09] CSeq: 120 NOTIFY [Jan 16 17:42:09] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:09] Subscription-State: active [Jan 16 17:42:09] Event: dialog [Jan 16 17:42:09] Content-Type: application/dialog-info+xml [Jan 16 17:42:09] Content-Length: 201 [Jan 16 17:42:09] [Jan 16 17:42:09] [Jan 16 17:42:09] [Jan 16 17:42:09] [Jan 16 17:42:09] confirmed [Jan 16 17:42:09] [Jan 16 17:42:09] [Jan 16 17:42:09] [Jan 16 17:42:09] --- [Jan 16 17:42:09] Really destroying SIP dialog 'd9c51afef5158d533b98edc24788a557' Method: INVITE [Jan 16 17:42:09] -- Playing 'digits/1.ulaw' (language 'en') [Jan 16 17:42:10] Retransmitting #3 (no NAT) to 192.168.0.50:5061: [Jan 16 17:42:10] OPTIONS sip:spa3102-pstn@192.168.0.50:5061 SIP/2.0 [Jan 16 17:42:10] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK46ae0331 [Jan 16 17:42:10] Max-Forwards: 70 [Jan 16 17:42:10] From: "asterisk" ;tag=as553c1f1a [Jan 16 17:42:10] To: [Jan 16 17:42:10] Contact: [Jan 16 17:42:10] Call-ID: 3a0a175e023a20284563641174710b53@192.168.0.20:5060 [Jan 16 17:42:10] CSeq: 102 OPTIONS [Jan 16 17:42:10] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:10] Date: Fri, 16 Jan 2015 16:42:07 GMT [Jan 16 17:42:10] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:10] Supported: replaces, timer [Jan 16 17:42:10] Content-Length: 0 [Jan 16 17:42:10] [Jan 16 17:42:10] [Jan 16 17:42:10] --- [Jan 16 17:42:10] -- Playing 'vm-INBOX.ulaw' (language 'en') [Jan 16 17:42:10] -- Playing 'vm-first.ulaw' (language 'en') [Jan 16 17:42:11] Retransmitting #4 (no NAT) to 192.168.0.50:5061: [Jan 16 17:42:11] OPTIONS sip:spa3102-pstn@192.168.0.50:5061 SIP/2.0 [Jan 16 17:42:11] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK46ae0331 [Jan 16 17:42:11] Max-Forwards: 70 [Jan 16 17:42:11] From: "asterisk" ;tag=as553c1f1a [Jan 16 17:42:11] To: [Jan 16 17:42:11] Contact: [Jan 16 17:42:11] Call-ID: 3a0a175e023a20284563641174710b53@192.168.0.20:5060 [Jan 16 17:42:11] CSeq: 102 OPTIONS [Jan 16 17:42:11] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:11] Date: Fri, 16 Jan 2015 16:42:07 GMT [Jan 16 17:42:11] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:11] Supported: replaces, timer [Jan 16 17:42:11] Content-Length: 0 [Jan 16 17:42:11] [Jan 16 17:42:11] [Jan 16 17:42:11] --- [Jan 16 17:42:11] Really destroying SIP dialog '3a0a175e023a20284563641174710b53@192.168.0.20:5060' Method: OPTIONS [Jan 16 17:42:11] Retransmitting #2 (no NAT) to 192.168.0.51:5060: [Jan 16 17:42:11] NOTIFY sip:76@192.168.0.51:5060 SIP/2.0 [Jan 16 17:42:11] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK6d032626;rport [Jan 16 17:42:11] Max-Forwards: 70 [Jan 16 17:42:11] From: ;tag=as02d09b91 [Jan 16 17:42:11] To: ;tag=sytb991fg3 [Jan 16 17:42:11] Contact: [Jan 16 17:42:11] Call-ID: 54b938661fd9-rg4njam42org [Jan 16 17:42:11] CSeq: 120 NOTIFY [Jan 16 17:42:11] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:11] Subscription-State: active [Jan 16 17:42:11] Event: dialog [Jan 16 17:42:11] Content-Type: application/dialog-info+xml [Jan 16 17:42:11] Content-Length: 201 [Jan 16 17:42:11] [Jan 16 17:42:11] [Jan 16 17:42:11] [Jan 16 17:42:11] [Jan 16 17:42:11] confirmed [Jan 16 17:42:11] [Jan 16 17:42:11] [Jan 16 17:42:11] [Jan 16 17:42:11] --- [Jan 16 17:42:11] == Parsing '/var/spool/asterisk/voicemail/default/800/INBOX/msg0000.txt': Found [Jan 16 17:42:11] -- Playing 'vm-message.ulaw' (language 'en') [Jan 16 17:42:11] Retransmitting #10 (no NAT) to 173.208.168.34:5070: [Jan 16 17:42:11] SIP/2.0 401 Unauthorized [Jan 16 17:42:11] Via: SIP/2.0/UDP 173.208.168.34:5070;branch=z9hG4bK-bce48f4927fd0834c36002e0cbc94cd7;received=173.208.168.34;rport=5070 [Jan 16 17:42:11] From: 2000;tag=18988e88 [Jan 16 17:42:11] To: 700441904891104;tag=as35cd4ca0 [Jan 16 17:42:11] Call-ID: bce48f4927fd0834c36002e0cbc94cd7 [Jan 16 17:42:11] CSeq: 1 INVITE [Jan 16 17:42:11] Server: Asterisk PBX 13.1.0 [Jan 16 17:42:11] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:11] Supported: replaces, timer [Jan 16 17:42:11] WWW-Authenticate: Digest algorithm=MD5, realm="kaworu.kunbox.net", nonce="3c54d5ea" [Jan 16 17:42:11] Content-Length: 0 [Jan 16 17:42:11] [Jan 16 17:42:11] [Jan 16 17:42:11] --- [Jan 16 17:42:12] == Spawn extension (SoftPhone, 98, 1) exited non-zero on 'SIP/76-0000006f' [Jan 16 17:42:12] Scheduling destruction of SIP dialog '54b93f5e0952-1zsswll6gkyr' in 6400 ms (Method: ACK) [Jan 16 17:42:12] set_destination: Parsing for address/port to send to [Jan 16 17:42:12] set_destination: set destination to 192.168.0.51:5060 [Jan 16 17:42:12] Reliably Transmitting (no NAT) to 192.168.0.51:5060: [Jan 16 17:42:12] BYE sip:76@192.168.0.51:5060 SIP/2.0 [Jan 16 17:42:12] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK2a3f2520;rport [Jan 16 17:42:12] Max-Forwards: 70 [Jan 16 17:42:12] From: ;tag=as4e86b1b1 [Jan 16 17:42:12] To: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:12] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:12] CSeq: 102 BYE [Jan 16 17:42:12] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:12] Proxy-Authorization: Digest username="76", realm="kaworu.kunbox.net", algorithm=MD5, uri="sip:192.168.0.20", nonce="53cb07ad", response="481ac53126969e56c41082b9e7373fee" [Jan 16 17:42:12] X-Asterisk-HangupCause: Unknown [Jan 16 17:42:12] X-Asterisk-HangupCauseCode: 0 [Jan 16 17:42:12] Content-Length: 0 [Jan 16 17:42:12] [Jan 16 17:42:12] [Jan 16 17:42:12] --- [Jan 16 17:42:12] set_destination: Parsing for address/port to send to [Jan 16 17:42:12] set_destination: set destination to 192.168.0.52:5060 [Jan 16 17:42:12] Reliably Transmitting (no NAT) to 192.168.0.52:5060: [Jan 16 17:42:12] NOTIFY sip:91@192.168.0.52:5060 SIP/2.0 [Jan 16 17:42:12] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK2e7dfba1;rport [Jan 16 17:42:12] Max-Forwards: 70 [Jan 16 17:42:12] From: ;tag=as1254995d [Jan 16 17:42:12] To: ;tag=5vsgks96rs [Jan 16 17:42:12] Contact: [Jan 16 17:42:12] Call-ID: 54b935d06aab-7wxqk99jossv [Jan 16 17:42:12] CSeq: 125 NOTIFY [Jan 16 17:42:12] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:12] Subscription-State: active [Jan 16 17:42:12] Event: dialog [Jan 16 17:42:12] Content-Type: application/dialog-info+xml [Jan 16 17:42:12] Content-Length: 202 [Jan 16 17:42:12] [Jan 16 17:42:12] [Jan 16 17:42:12] [Jan 16 17:42:12] [Jan 16 17:42:12] terminated [Jan 16 17:42:12] [Jan 16 17:42:12] [Jan 16 17:42:12] [Jan 16 17:42:12] --- [Jan 16 17:42:12] == Extension Changed 76[SoftPhone] new state Idle for Notify User 91 [Jan 16 17:42:12] == Extension Changed 76[SoftPhone] new state Idle for Notify User 76 (queued) [Jan 16 17:42:12] [Jan 16 17:42:12] <--- SIP read from UDP:192.168.0.52:5060 ---> [Jan 16 17:42:12] SIP/2.0 200 Ok [Jan 16 17:42:12] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK2e7dfba1;rport=5060 [Jan 16 17:42:12] From: ;tag=as1254995d [Jan 16 17:42:12] To: ;tag=5vsgks96rs [Jan 16 17:42:12] Call-ID: 54b935d06aab-7wxqk99jossv [Jan 16 17:42:12] CSeq: 125 NOTIFY [Jan 16 17:42:12] Content-Length: 0 [Jan 16 17:42:12] [Jan 16 17:42:12] <-------------> [Jan 16 17:42:12] --- (7 headers 0 lines) --- [Jan 16 17:42:12] Retransmitting #1 (no NAT) to 192.168.0.51:5060: [Jan 16 17:42:12] BYE sip:76@192.168.0.51:5060 SIP/2.0 [Jan 16 17:42:12] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK2a3f2520;rport [Jan 16 17:42:12] Max-Forwards: 70 [Jan 16 17:42:12] From: ;tag=as4e86b1b1 [Jan 16 17:42:12] To: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:12] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:12] CSeq: 102 BYE [Jan 16 17:42:12] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:12] Proxy-Authorization: Digest username="76", realm="kaworu.kunbox.net", algorithm=MD5, uri="sip:192.168.0.20", nonce="53cb07ad", response="481ac53126969e56c41082b9e7373fee" [Jan 16 17:42:12] X-Asterisk-HangupCause: Unknown [Jan 16 17:42:12] X-Asterisk-HangupCauseCode: 0 [Jan 16 17:42:12] Content-Length: 0 [Jan 16 17:42:12] [Jan 16 17:42:12] [Jan 16 17:42:12] --- [Jan 16 17:42:12] [Jan 16 17:42:12] <--- SIP read from UDP:192.168.0.51:5060 ---> [Jan 16 17:42:12] SIP/2.0 200 OK [Jan 16 17:42:12] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK2a3f2520;rport=5060 [Jan 16 17:42:12] From: ;tag=as4e86b1b1 [Jan 16 17:42:12] To: "Wohnzimmer" ;tag=zb96jyz2p0 [Jan 16 17:42:12] Call-ID: 54b93f5e0952-1zsswll6gkyr [Jan 16 17:42:12] CSeq: 102 BYE [Jan 16 17:42:12] Contact: ;reg-id=1 [Jan 16 17:42:12] User-Agent: snom360/8.7.3.25 [Jan 16 17:42:12] RTP-RxStat: Total_Rx_Pkts=179,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 [Jan 16 17:42:12] RTP-TxStat: Total_Tx_Pkts=181,Tx_Pkts=181,Remote_Tx_Pkts=0 [Jan 16 17:42:12] Content-Length: 0 [Jan 16 17:42:12] [Jan 16 17:42:12] <-------------> [Jan 16 17:42:12] --- (11 headers 0 lines) --- [Jan 16 17:42:12] SIP Response message for INCOMING dialog BYE arrived [Jan 16 17:42:12] Really destroying SIP dialog '54b93f5e0952-1zsswll6gkyr' Method: ACK [Jan 16 17:42:15] Retransmitting #3 (no NAT) to 192.168.0.51:5060: [Jan 16 17:42:15] NOTIFY sip:76@192.168.0.51:5060 SIP/2.0 [Jan 16 17:42:15] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK6d032626;rport [Jan 16 17:42:15] Max-Forwards: 70 [Jan 16 17:42:15] From: ;tag=as02d09b91 [Jan 16 17:42:15] To: ;tag=sytb991fg3 [Jan 16 17:42:15] Contact: [Jan 16 17:42:15] Call-ID: 54b938661fd9-rg4njam42org [Jan 16 17:42:15] CSeq: 120 NOTIFY [Jan 16 17:42:15] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:15] Subscription-State: active [Jan 16 17:42:15] Event: dialog [Jan 16 17:42:15] Content-Type: application/dialog-info+xml [Jan 16 17:42:15] Content-Length: 201 [Jan 16 17:42:15] [Jan 16 17:42:15] [Jan 16 17:42:15] [Jan 16 17:42:15] [Jan 16 17:42:15] confirmed [Jan 16 17:42:15] [Jan 16 17:42:15] [Jan 16 17:42:15] [Jan 16 17:42:15] --- [Jan 16 17:42:15] [Jan 16 17:42:15] <--- SIP read from UDP:192.168.0.51:5060 ---> [Jan 16 17:42:15] SIP/2.0 200 Ok [Jan 16 17:42:15] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK6d032626;rport=5060 [Jan 16 17:42:15] From: ;tag=as02d09b91 [Jan 16 17:42:15] To: ;tag=sytb991fg3 [Jan 16 17:42:15] Call-ID: 54b938661fd9-rg4njam42org [Jan 16 17:42:15] CSeq: 120 NOTIFY [Jan 16 17:42:15] Content-Length: 0 [Jan 16 17:42:15] [Jan 16 17:42:15] <-------------> [Jan 16 17:42:15] --- (7 headers 0 lines) --- [Jan 16 17:42:15] set_destination: Parsing for address/port to send to [Jan 16 17:42:15] set_destination: set destination to 192.168.0.51:5060 [Jan 16 17:42:15] Reliably Transmitting (no NAT) to 192.168.0.51:5060: [Jan 16 17:42:15] NOTIFY sip:76@192.168.0.51:5060 SIP/2.0 [Jan 16 17:42:15] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK2f1a50df;rport [Jan 16 17:42:15] Max-Forwards: 70 [Jan 16 17:42:15] From: ;tag=as02d09b91 [Jan 16 17:42:15] To: ;tag=sytb991fg3 [Jan 16 17:42:15] Contact: [Jan 16 17:42:15] Call-ID: 54b938661fd9-rg4njam42org [Jan 16 17:42:15] CSeq: 121 NOTIFY [Jan 16 17:42:15] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:15] Subscription-State: active [Jan 16 17:42:15] Event: dialog [Jan 16 17:42:15] Content-Type: application/dialog-info+xml [Jan 16 17:42:15] Content-Length: 202 [Jan 16 17:42:15] [Jan 16 17:42:15] [Jan 16 17:42:15] [Jan 16 17:42:15] [Jan 16 17:42:15] terminated [Jan 16 17:42:15] [Jan 16 17:42:15] [Jan 16 17:42:15] [Jan 16 17:42:15] --- [Jan 16 17:42:16] Retransmitting #1 (no NAT) to 192.168.0.51:5060: [Jan 16 17:42:16] NOTIFY sip:76@192.168.0.51:5060 SIP/2.0 [Jan 16 17:42:16] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK2f1a50df;rport [Jan 16 17:42:16] Max-Forwards: 70 [Jan 16 17:42:16] From: ;tag=as02d09b91 [Jan 16 17:42:16] To: ;tag=sytb991fg3 [Jan 16 17:42:16] Contact: [Jan 16 17:42:16] Call-ID: 54b938661fd9-rg4njam42org [Jan 16 17:42:16] CSeq: 121 NOTIFY [Jan 16 17:42:16] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:16] Subscription-State: active [Jan 16 17:42:16] Event: dialog [Jan 16 17:42:16] Content-Type: application/dialog-info+xml [Jan 16 17:42:16] Content-Length: 202 [Jan 16 17:42:16] [Jan 16 17:42:16] [Jan 16 17:42:16] [Jan 16 17:42:16] [Jan 16 17:42:16] terminated [Jan 16 17:42:16] [Jan 16 17:42:16] [Jan 16 17:42:16] [Jan 16 17:42:16] --- [Jan 16 17:42:16] [Jan 16 17:42:16] <--- SIP read from UDP:192.168.0.51:5060 ---> [Jan 16 17:42:16] SIP/2.0 200 Ok [Jan 16 17:42:16] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK2f1a50df;rport=5060 [Jan 16 17:42:16] From: ;tag=as02d09b91 [Jan 16 17:42:16] To: ;tag=sytb991fg3 [Jan 16 17:42:16] Call-ID: 54b938661fd9-rg4njam42org [Jan 16 17:42:16] CSeq: 121 NOTIFY [Jan 16 17:42:16] Content-Length: 0 [Jan 16 17:42:16] [Jan 16 17:42:16] <-------------> [Jan 16 17:42:16] --- (7 headers 0 lines) --- [Jan 16 17:42:18] Reliably Transmitting (no NAT) to 192.168.0.180:62214: [Jan 16 17:42:18] OPTIONS sip:handy-felix@192.168.0.180:62214;app-id=org.linphone.phone.prod;pn-type=apple;pn-tok=1A4629A367797941D497C8662E11391B2C79C574B4698F6C723A72CBD140A4FF;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf SIP/2.0 [Jan 16 17:42:18] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK037738f7 [Jan 16 17:42:18] Max-Forwards: 70 [Jan 16 17:42:18] From: "asterisk" ;tag=as5eb09f57 [Jan 16 17:42:18] To: [Jan 16 17:42:18] Contact: [Jan 16 17:42:18] Call-ID: 39690ff800bf4e8b30aa573f714a6947@192.168.0.20:5060 [Jan 16 17:42:18] CSeq: 102 OPTIONS [Jan 16 17:42:18] User-Agent: Asterisk PBX 13.1.0 [Jan 16 17:42:18] Date: Fri, 16 Jan 2015 16:42:18 GMT [Jan 16 17:42:18] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 16 17:42:18] Supported: replaces, timer [Jan 16 17:42:18] Content-Length: 0 [Jan 16 17:42:18] [Jan 16 17:42:18] [Jan 16 17:42:18] --- kaworu*CLI> sip set debug off SIP Debugging Disabled