SIP Debugging Enabled for IP: 192.168.1.65:5060 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=z9hG4bK439f8993dd585945c.9dead3608f72589dd Max-Forwards: 70 From: "T1902 Receptie2" ;tag=0894099b01 To: "009000909" Call-ID: e71b2f96a931bf99 CSeq: 28771 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: 227 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 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 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 - e71b2f96a931bf99 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 audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 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 - 0x0 (nothing), combined - 0x0 (nothing) 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: <--- 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=z9hG4bK439f8993dd585945c.9dead3608f72589dd;received=192.168.1.65 From: "T1902 Receptie2" ;tag=0894099b01 To: "009000909" Call-ID: e71b2f96a931bf99 CSeq: 28771 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=z9hG4bK4f4037bd;rport Max-Forwards: 70 From: "" ;tag=as20e1bfa6 To: "T1902 Receptie2" ;tag=0d0ec441f9 Contact: Call-ID: 1cd9c863daf3a20e CSeq: 106 NOTIFY User-Agent: Asterisk PBX 1.6.1.11 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 209 --- == Extension Changed 1902[blf] new state Busy for Notify User 1902 == Extension Changed 1902[blf] new state Busy for Notify User 1901 -- 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=z9hG4bK4f4037bd;rport=5060;received=192.168.3.18 From: "" ;tag=as20e1bfa6 To: "T1902 Receptie2" ;tag=0d0ec441f9 Call-ID: 1cd9c863daf3a20e CSeq: 106 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 15128 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP wglpbx*CLI> <--- 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=z9hG4bK439f8993dd585945c.9dead3608f72589dd;received=192.168.1.65 From: "T1902 Receptie2" ;tag=0894099b01 To: "009000909" ;tag=as3792692d Call-ID: e71b2f96a931bf99 CSeq: 28771 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: 279 v=0 o=root 1138646646 1138646646 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 15128 RTP/AVP 8 0 18 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> 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=z9hG4bKca330b1efdbafef6d.f482f44e5695bad79 Max-Forwards: 70 From: ;tag=268679d153 To: Call-ID: 585f42bfbeeb575a CSeq: 16114 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="34f5c54f",uri="sip:192.168.3.18:5060",response="8df02fef9c72c56d96cc18f0417075d2",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 18 10:58:37] NOTICE[3442]: chan_sip.c:11847 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=z9hG4bKca330b1efdbafef6d.f482f44e5695bad79;received=192.168.1.65 From: ;tag=268679d153 To: ;tag=as591d7eb4 Call-ID: 585f42bfbeeb575a CSeq: 16114 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="5ab6bd48", stale=true Content-Length: 0 <------------> Scheduling destruction of SIP dialog '585f42bfbeeb575a' 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=z9hG4bK7a9aae520fc6f88e5.e13bcc708f5cae094 Max-Forwards: 70 From: ;tag=268679d153 To: Call-ID: 585f42bfbeeb575a CSeq: 16115 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="5ab6bd48",uri="sip:192.168.3.18:5060",response="fc89b9d7470d2a0cee6991996297fd9c",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=z9hG4bK7a9aae520fc6f88e5.e13bcc708f5cae094;received=192.168.1.65 From: ;tag=268679d153 To: ;tag=as591d7eb4 Call-ID: 585f42bfbeeb575a CSeq: 16115 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: Fri, 18 Dec 2009 09:58:37 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '585f42bfbeeb575a' in 32000 ms (Method: REGISTER) -- 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 15128 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) 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=z9hG4bK439f8993dd585945c.9dead3608f72589dd;received=192.168.1.65 From: "T1902 Receptie2" ;tag=0894099b01 To: "009000909" ;tag=as3792692d Call-ID: e71b2f96a931bf99 CSeq: 28771 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: 279 v=0 o=root 1138646646 1138646647 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 15128 RTP/AVP 8 0 18 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no 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=z9hG4bKaad3d45abbd9a3500.32907478e077c3b68 Max-Forwards: 70 From: "T1902 Receptie2" ;tag=0894099b01 To: "009000909" ;tag=as3792692d Call-ID: e71b2f96a931bf99 CSeq: 28771 ACK User-Agent: Aastra 6731i/2.5.2.1010 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Dec 18 10:58:45] NOTICE[3442]: chan_sip.c:11847 check_auth: Correct auth, but based on stale nonce received from '' [Dec 18 10:58:45] NOTICE[3442]: chan_sip.c:11847 check_auth: Correct auth, but based on stale nonce received from '' 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=z9hG4bKb8159540a8132b8e4.11eca7c97080db806 Max-Forwards: 70 From: "T1902 Receptie2" ;tag=0894099b01 To: "009000909" ;tag=as3792692d Call-ID: e71b2f96a931bf99 CSeq: 28772 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=z9hG4bKb8159540a8132b8e4.11eca7c97080db806;received=192.168.1.65 From: "T1902 Receptie2" ;tag=0894099b01 To: "009000909" ;tag=as3792692d Call-ID: e71b2f96a931bf99 CSeq: 28772 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=z9hG4bK6d87032b;rport Max-Forwards: 70 From: "" ;tag=as20e1bfa6 To: "T1902 Receptie2" ;tag=0d0ec441f9 Contact: Call-ID: 1cd9c863daf3a20e CSeq: 107 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 Idle for Notify User 1902 == Extension Changed 1902[blf] new state Idle for Notify User 1901 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=z9hG4bK6d87032b;rport=5060;received=192.168.3.18 From: "" ;tag=as20e1bfa6 To: "T1902 Receptie2" ;tag=0d0ec441f9 Call-ID: 1cd9c863daf3a20e CSeq: 107 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 Really destroying SIP dialog 'e71b2f96a931bf99' Method: BYE 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=z9hG4bK0ad6feee9209e64d2.d8bf14c99ce1d5fb9 Max-Forwards: 70 From: ;tag=268679d153 To: Call-ID: 585f42bfbeeb575a CSeq: 16116 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="5ab6bd48",uri="sip:192.168.3.18:5060",response="fc89b9d7470d2a0cee6991996297fd9c",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 18 10:58:52] NOTICE[3442]: chan_sip.c:11847 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=z9hG4bK0ad6feee9209e64d2.d8bf14c99ce1d5fb9;received=192.168.1.65 From: ;tag=268679d153 To: ;tag=as591d7eb4 Call-ID: 585f42bfbeeb575a CSeq: 16116 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="04f4fae2", stale=true Content-Length: 0 <------------> Scheduling destruction of SIP dialog '585f42bfbeeb575a' 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=z9hG4bKacb92cd339ce3b349.7239474e2f7b164ee Max-Forwards: 70 From: ;tag=268679d153 To: Call-ID: 585f42bfbeeb575a CSeq: 16117 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="04f4fae2",uri="sip:192.168.3.18:5060",response="c80b6ee0ad76485d767295867f6cc57d",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=z9hG4bKacb92cd339ce3b349.7239474e2f7b164ee;received=192.168.1.65 From: ;tag=268679d153 To: ;tag=as591d7eb4 Call-ID: 585f42bfbeeb575a CSeq: 16117 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: Fri, 18 Dec 2009 09:58:52 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '585f42bfbeeb575a' in 32000 ms (Method: REGISTER) wglpbx*CLI> sip set debug off SIP Debugging Disabled