SIP Debugging Enabled for IP: 192.168.1.65:5060 wglpbx*CLI> <--- SIP read from UDP://192.168.1.65:5060 ---> SUBSCRIBE sip:3002@192.168.3.18 SIP/2.0 Accept: application/dialog-info+xml Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK1812ffa46fe74604a.a37a6fdf4c7903836 Max-Forwards: 70 From: "T1902 Receptie2" ;tag=a4d4f583fc To: "" ;tag=as315a07bb Call-ID: 430968cb7ff2307f CSeq: 4767 SUBSCRIBE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "T1902 Receptie2" ;+sip.instance="" Event: dialog Expires: 120 Supported: gruu, path User-Agent: Aastra 6731i/2.5.2.1010 Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Found peer '1902' for '1902' from 192.168.1.65:5060 Looking for 3002 in asn-park (domain 192.168.3.18) Scheduling destruction of SIP dialog '430968cb7ff2307f' in 130000 ms (Method: SUBSCRIBE) <--- Transmitting (no NAT) to 192.168.1.65:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK1812ffa46fe74604a.a37a6fdf4c7903836;received=192.168.1.65 From: "T1902 Receptie2" ;tag=a4d4f583fc To: "" ;tag=as315a07bb Call-ID: 430968cb7ff2307f CSeq: 4767 SUBSCRIBE Server: Asterisk PBX 1.6.1.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 120 Contact: ;expires=120 Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.65, port 5060 Reliably Transmitting (no NAT) to 192.168.1.65:5060: NOTIFY sip:1902@192.168.1.65:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.3.18:5060;branch=z9hG4bK1bf2694a;rport Max-Forwards: 70 From: "" ;tag=as315a07bb To: "T1902 Receptie2" ;tag=a4d4f583fc Contact: Call-ID: 430968cb7ff2307f CSeq: 180 NOTIFY User-Agent: Asterisk PBX 1.6.1.11 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 211 --- wglpbx*CLI> <--- SIP read from UDP://192.168.1.65:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.18:5060;branch=z9hG4bK1bf2694a;rport=5060;received=192.168.3.18 From: "" ;tag=as315a07bb To: "T1902 Receptie2" ;tag=a4d4f583fc Call-ID: 430968cb7ff2307f CSeq: 180 NOTIFY Contact: "T1902 Receptie2" ;+sip.instance="" Server: Aastra 6731i/2.5.2.1010 Content-Length: 0 wglpbx*CLI> <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived wglpbx*CLI> <--- SIP read from UDP://192.168.1.65:5060 ---> INVITE sip:009000909@192.168.3.18:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bKc49c4be7199862f21.d1de95dbd504b7781 Max-Forwards: 70 From: "T1902 Receptie2" ;tag=3b728fd5c3 To: "009000909" Call-ID: 1d2ce4759fb5848c CSeq: 27831 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "T1902 Receptie2" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 6731i/2.5.2.1010 Content-Type: application/sdp Content-Length: 283 v=0 o=MxSIP 0 0 IN IP4 192.168.1.65 s=SIP Call c=IN IP4 192.168.1.65 t=0 0 m=audio 3000 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (14 headers 14 lines) --- == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 Sending to 192.168.1.65 : 5060 (no NAT) Using INVITE request as basis request - 1d2ce4759fb5848c Found peer '1902' for '1902' from 192.168.1.65:5060 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.65:3000 Looking for 009000909 in asn-receptie (domain 192.168.3.18) list_route: hop: wglpbx*CLI> <--- Transmitting (no NAT) to 192.168.1.65:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bKc49c4be7199862f21.d1de95dbd504b7781;received=192.168.1.65 From: "T1902 Receptie2" ;tag=3b728fd5c3 To: "009000909" Call-ID: 1d2ce4759fb5848c CSeq: 27831 INVITE Server: Asterisk PBX 1.6.1.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.65, port 5060 Reliably Transmitting (no NAT) to 192.168.1.65:5060: NOTIFY sip:1902@192.168.1.65:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.3.18:5060;branch=z9hG4bK1a3b190d;rport Max-Forwards: 70 From: "" ;tag=as34824ced To: "T1902 Receptie2" ;tag=5d563fe782 Contact: Call-ID: 3c35b085fac7d981 CSeq: 192 NOTIFY User-Agent: Asterisk PBX 1.6.1.11 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 210 --- == Extension Changed 1902[blf] new state Busy for Notify User 1902 -- Executing [009000909@asn-receptie:1] Dial("SIP/1902-0000000e", "SIP/Priority_out/09000909,,T") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 -- Called Priority_out/09000909 wglpbx*CLI> <--- SIP read from UDP://192.168.1.65:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.18:5060;branch=z9hG4bK1a3b190d;rport=5060;received=192.168.3.18 From: "" ;tag=as34824ced To: "T1902 Receptie2" ;tag=5d563fe782 Call-ID: 3c35b085fac7d981 CSeq: 192 NOTIFY Contact: "T1902 Receptie2" ;+sip.instance="" Server: Aastra 6731i/2.5.2.1010 Content-Length: 0 wglpbx*CLI> <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- SIP/Priority_out-0000000f is making progress passing it to SIP/1902-0000000e Audio is at 192.168.3.18 port 17422 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.1.65:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bKc49c4be7199862f21.d1de95dbd504b7781;received=192.168.1.65 From: "T1902 Receptie2" ;tag=3b728fd5c3 To: "009000909" ;tag=as3921ded1 Call-ID: 1d2ce4759fb5848c CSeq: 27831 INVITE Server: Asterisk PBX 1.6.1.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 335 v=0 o=root 1627218371 1627218371 IN IP4 192.168.3.18 s=Asterisk PBX 1.6.1.11 c=IN IP4 192.168.3.18 t=0 0 m=audio 17422 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- SIP/Priority_out-0000000f is making progress passing it to SIP/1902-0000000e -- SIP/Priority_out-0000000f answered SIP/1902-0000000e Audio is at 192.168.3.18 port 17422 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.1.65:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bKc49c4be7199862f21.d1de95dbd504b7781;received=192.168.1.65 From: "T1902 Receptie2" ;tag=3b728fd5c3 To: "009000909" ;tag=as3921ded1 Call-ID: 1d2ce4759fb5848c CSeq: 27831 INVITE Server: Asterisk PBX 1.6.1.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 335 v=0 o=root 1627218371 1627218372 IN IP4 192.168.3.18 s=Asterisk PBX 1.6.1.11 c=IN IP4 192.168.3.18 t=0 0 m=audio 17422 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> wglpbx*CLI> <--- SIP read from UDP://192.168.1.65:5060 ---> ACK sip:009000909@192.168.3.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK0530bf8500befeb2a.baed41a6c85af09f5 Max-Forwards: 70 From: "T1902 Receptie2" ;tag=3b728fd5c3 To: "009000909" ;tag=as3921ded1 Call-ID: 1d2ce4759fb5848c CSeq: 27831 ACK User-Agent: Aastra 6731i/2.5.2.1010 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Dec 14 16:40:08] DTMF[26583]: channel.c:2846 __ast_read: DTMF begin '3' received on SIP/1902-0000000e [Dec 14 16:40:08] DTMF[26583]: channel.c:2856 __ast_read: DTMF begin passthrough '3' on SIP/1902-0000000e [Dec 14 16:40:08] DTMF[26583]: channel.c:2774 __ast_read: DTMF end '3' received on SIP/1902-0000000e, duration 110 ms [Dec 14 16:40:08] DTMF[26583]: channel.c:2814 __ast_read: DTMF end accepted with begin '3' on SIP/1902-0000000e [Dec 14 16:40:08] DTMF[26583]: channel.c:2830 __ast_read: DTMF end passthrough '3' on SIP/1902-0000000e [Dec 14 16:40:08] DTMF[26519]: channel.c:2846 __ast_read: DTMF begin '4' received on SIP/1902-0000000e [Dec 14 16:40:08] DTMF[26519]: channel.c:2850 __ast_read: DTMF begin ignored '4' on SIP/1902-0000000e [Dec 14 16:40:08] DTMF[26583]: channel.c:2774 __ast_read: DTMF end '4' received on SIP/1902-0000000e, duration 110 ms [Dec 14 16:40:08] DTMF[26583]: channel.c:2800 __ast_read: DTMF begin emulation of '4' with duration 110 queued on SIP/1902-0000000e [Dec 14 16:40:08] DTMF[26583]: channel.c:2923 __ast_read: DTMF end emulation of '4' queued on SIP/1902-0000000e [Dec 14 16:40:08] DTMF[26519]: channel.c:2846 __ast_read: DTMF begin '6' received on SIP/1902-0000000e [Dec 14 16:40:08] DTMF[26519]: channel.c:2850 __ast_read: DTMF begin ignored '6' on SIP/1902-0000000e [Dec 14 16:40:09] DTMF[26583]: channel.c:2774 __ast_read: DTMF end '6' received on SIP/1902-0000000e, duration 120 ms [Dec 14 16:40:09] DTMF[26583]: channel.c:2800 __ast_read: DTMF begin emulation of '6' with duration 120 queued on SIP/1902-0000000e [Dec 14 16:40:09] DTMF[26583]: channel.c:2846 __ast_read: DTMF begin '9' received on SIP/1902-0000000e [Dec 14 16:40:09] DTMF[26583]: channel.c:2850 __ast_read: DTMF begin ignored '9' on SIP/1902-0000000e [Dec 14 16:40:09] DTMF[26583]: channel.c:2879 __ast_read: DTMF end emulation of '6' queued on SIP/1902-0000000e [Dec 14 16:40:09] DTMF[26519]: channel.c:2774 __ast_read: DTMF end '9' received on SIP/1902-0000000e, duration 110 ms [Dec 14 16:40:09] DTMF[26519]: channel.c:2830 __ast_read: DTMF end passthrough '9' on SIP/1902-0000000e wglpbx*CLI> <--- SIP read from UDP://192.168.1.65:5060 ---> REGISTER sip:192.168.3.18:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK4b4b8d0e2e14fe7ac.7d7c2beab7f744a1d Max-Forwards: 70 From: ;tag=9201fb26ae To: Call-ID: 0d5e8ab61a60931c CSeq: 27491 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="1902",realm="asterisk",nonce="0168666a",uri="sip:192.168.3.18:5060",response="e57591cf027684dbfb5f24e698ac4534",algorithm=MD5 Contact: T1902 Receptie2 ;expires=30;+sip.instance="" Supported: gruu, path User-Agent: Aastra 6731i/2.5.2.1010 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 192.168.1.65 : 5060 (no NAT) [Dec 14 16:40:09] NOTICE[26496]: chan_sip.c:11854 check_auth: Correct auth, but based on stale nonce received from '' <--- Transmitting (no NAT) to 192.168.1.65:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK4b4b8d0e2e14fe7ac.7d7c2beab7f744a1d;received=192.168.1.65 From: ;tag=9201fb26ae To: ;tag=as58ed90b7 Call-ID: 0d5e8ab61a60931c CSeq: 27491 REGISTER Server: Asterisk PBX 1.6.1.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d260c7d", stale=true Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0d5e8ab61a60931c' in 32000 ms (Method: REGISTER) wglpbx*CLI> <--- SIP read from UDP://192.168.1.65:5060 ---> REGISTER sip:192.168.3.18:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bKf761ad5039fd15201.01f053da146400fc8 Max-Forwards: 70 From: ;tag=9201fb26ae To: Call-ID: 0d5e8ab61a60931c CSeq: 27492 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="1902",realm="asterisk",nonce="3d260c7d",uri="sip:192.168.3.18:5060",response="e60923af88c44fe64caa7a7748959dcf",algorithm=MD5 Contact: T1902 Receptie2 ;expires=30;+sip.instance="" Supported: gruu, path User-Agent: Aastra 6731i/2.5.2.1010 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 192.168.1.65 : 5060 (no NAT) wglpbx*CLI> <--- Transmitting (no NAT) to 192.168.1.65:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bKf761ad5039fd15201.01f053da146400fc8;received=192.168.1.65 From: ;tag=9201fb26ae To: ;tag=as58ed90b7 Call-ID: 0d5e8ab61a60931c CSeq: 27492 REGISTER Server: Asterisk PBX 1.6.1.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 30 Contact: sip:1902@192.168.1.65:5060;transport=udp;expires=30 Date: Mon, 14 Dec 2009 15:40:09 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0d5e8ab61a60931c' in 32000 ms (Method: REGISTER) [Dec 14 16:40:09] DTMF[26583]: channel.c:2774 __ast_read: DTMF end '9' received on SIP/1902-0000000e, duration 110 ms [Dec 14 16:40:09] DTMF[26583]: channel.c:2800 __ast_read: DTMF begin emulation of '9' with duration 110 queued on SIP/1902-0000000e [Dec 14 16:40:09] DTMF[26583]: channel.c:2846 __ast_read: DTMF begin '9' received on SIP/1902-0000000e [Dec 14 16:40:09] DTMF[26583]: channel.c:2850 __ast_read: DTMF begin ignored '9' on SIP/1902-0000000e [Dec 14 16:40:09] DTMF[26583]: channel.c:2879 __ast_read: DTMF end emulation of '9' queued on SIP/1902-0000000e [Dec 14 16:40:09] DTMF[26519]: channel.c:2774 __ast_read: DTMF end '9' received on SIP/1902-0000000e, duration 110 ms [Dec 14 16:40:09] DTMF[26519]: channel.c:2830 __ast_read: DTMF end passthrough '9' on SIP/1902-0000000e [Dec 14 16:40:09] DTMF[26583]: channel.c:2774 __ast_read: DTMF end '9' received on SIP/1902-0000000e, duration 110 ms [Dec 14 16:40:09] DTMF[26583]: channel.c:2800 __ast_read: DTMF begin emulation of '9' with duration 110 queued on SIP/1902-0000000e [Dec 14 16:40:09] DTMF[26583]: channel.c:2923 __ast_read: DTMF end emulation of '9' queued on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26583]: channel.c:2846 __ast_read: DTMF begin '7' received on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26583]: channel.c:2856 __ast_read: DTMF begin passthrough '7' on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26583]: channel.c:2774 __ast_read: DTMF end '7' received on SIP/1902-0000000e, duration 80 ms [Dec 14 16:40:10] DTMF[26583]: channel.c:2814 __ast_read: DTMF end accepted with begin '7' on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26583]: channel.c:2830 __ast_read: DTMF end passthrough '7' on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26519]: channel.c:2846 __ast_read: DTMF begin '6' received on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26519]: channel.c:2850 __ast_read: DTMF begin ignored '6' on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26583]: channel.c:2774 __ast_read: DTMF end '6' received on SIP/1902-0000000e, duration 80 ms [Dec 14 16:40:10] DTMF[26583]: channel.c:2800 __ast_read: DTMF begin emulation of '6' with duration 80 queued on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26583]: channel.c:2923 __ast_read: DTMF end emulation of '6' queued on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26519]: channel.c:2846 __ast_read: DTMF begin '8' received on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26519]: channel.c:2850 __ast_read: DTMF begin ignored '8' on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26519]: channel.c:2774 __ast_read: DTMF end '8' received on SIP/1902-0000000e, duration 80 ms [Dec 14 16:40:10] DTMF[26519]: channel.c:2830 __ast_read: DTMF end passthrough '8' on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26583]: channel.c:2774 __ast_read: DTMF end '8' received on SIP/1902-0000000e, duration 80 ms [Dec 14 16:40:10] DTMF[26583]: channel.c:2800 __ast_read: DTMF begin emulation of '8' with duration 80 queued on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26583]: channel.c:2846 __ast_read: DTMF begin '2' received on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26583]: channel.c:2850 __ast_read: DTMF begin ignored '2' on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26583]: channel.c:2879 __ast_read: DTMF end emulation of '8' queued on SIP/1902-0000000e [Dec 14 16:40:10] DTMF[26519]: channel.c:2774 __ast_read: DTMF end '2' received on SIP/1902-0000000e, duration 100 ms [Dec 14 16:40:10] DTMF[26519]: channel.c:2830 __ast_read: DTMF end passthrough '2' on SIP/1902-0000000e [Dec 14 16:40:11] DTMF[26583]: channel.c:2774 __ast_read: DTMF end '2' received on SIP/1902-0000000e, duration 100 ms [Dec 14 16:40:11] DTMF[26583]: channel.c:2800 __ast_read: DTMF begin emulation of '2' with duration 100 queued on SIP/1902-0000000e [Dec 14 16:40:11] DTMF[26583]: channel.c:2923 __ast_read: DTMF end emulation of '2' queued on SIP/1902-0000000e wglpbx*CLI> <--- SIP read from UDP://192.168.1.65:5060 ---> BYE sip:009000909@192.168.3.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bKac326d7d103f4b97f.b7cc26557644c3077 Max-Forwards: 70 From: "T1902 Receptie2" ;tag=3b728fd5c3 To: "009000909" ;tag=as3921ded1 Call-ID: 1d2ce4759fb5848c CSeq: 27832 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Supported: gruu, path, timer User-Agent: Aastra 6731i/2.5.2.1010 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.1.65 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.1.65:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bKac326d7d103f4b97f.b7cc26557644c3077;received=192.168.1.65 From: "T1902 Receptie2" ;tag=3b728fd5c3 To: "009000909" ;tag=as3921ded1 Call-ID: 1d2ce4759fb5848c CSeq: 27832 BYE Server: Asterisk PBX 1.6.1.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (asn-receptie, 009000909, 1) exited non-zero on 'SIP/1902-0000000e' set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.65, port 5060 Reliably Transmitting (no NAT) to 192.168.1.65:5060: NOTIFY sip:1902@192.168.1.65:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.3.18:5060;branch=z9hG4bK5eb683a6;rport Max-Forwards: 70 From: "" ;tag=as34824ced To: "T1902 Receptie2" ;tag=5d563fe782 Contact: Call-ID: 3c35b085fac7d981 CSeq: 193 NOTIFY User-Agent: Asterisk PBX 1.6.1.11 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 211 --- == Extension Changed 1902[blf] new state Idle for Notify User 1902 Really destroying SIP dialog '1d2ce4759fb5848c' Method: BYE wglpbx*CLI> <--- SIP read from UDP://192.168.1.65:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.18:5060;branch=z9hG4bK5eb683a6;rport=5060;received=192.168.3.18 From: "" ;tag=as34824ced To: "T1902 Receptie2" ;tag=5d563fe782 Call-ID: 3c35b085fac7d981 CSeq: 193 NOTIFY Contact: "T1902 Receptie2" ;+sip.instance="" Server: Aastra 6731i/2.5.2.1010 Content-Length: 0 wglpbx*CLI> <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived