Asterisk SVN-trunk-r180641, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk SVN-trunk-r180641 currently running on LAsterisco (pid = 8776) LAsterisco*CLI> <--- SIP read from UDP:10.0.0.102:51858 ---> INVITE sip:1999@10.0.2.7 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.102:51858;branch=z9hG4bK-d8754z-d61ce123ac58964d-1---d8754z-;rport Max-Forwards: 70 Contact: To: "1999" From: "1012";tag=a7316a35 Call-ID: MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 874 v=0 o=- 3 2 IN IP4 10.0.0.102 s=CounterPath X-Lite 3.0 c=IN IP4 10.0.0.102 t=0 0 m=audio 48486 RTP/AVP 107 119 100 106 0 105 98 8 3 101 a=alt:1 10 : kF2I1u+G Sq4/O0pW 10.0.0.102 48486 a=alt:2 9 : KWRNSyiG VvDtHmYr 192.168.1.69 48486 a=alt:3 8 : LBz3mYzX uZQc8jSV 192.168.2.166 48486 a=alt:4 7 : VZMcem8i /4VgK0TP 192.168.2.11 48486 a=alt:5 6 : +pMMAjkA qVmyVw4Y 192.168.0.169 48486 a=alt:6 5 : h3/G7t2Q 3wwi7QBK 192.168.8.164 48486 a=alt:7 4 : W6SmDyuj Uf0tNweB 169.254.1.58 48486 a=alt:8 3 : IyNxodns EXWVfJ7a 192.168.222.69 48486 a=alt:9 2 : qPAh2cuh BJW1ixVZ 192.168.136.1 48486 a=alt:10 1 : 3s9JLXyc OK2aOwkp 192.168.56.1 48486 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (12 headers 25 lines) --- Sending to 10.0.0.102 : 51858 (NAT) Using INVITE request as basis request - MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. Found peer '1012' for '1012' from 10.0.0.102:51858 <--- Reliably Transmitting (no NAT) to 10.0.0.102:51858 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.102:51858;branch=z9hG4bK-d8754z-d61ce123ac58964d-1---d8754z-;received=10.0.0.102;rport=51858 From: "1012";tag=a7316a35 To: "1999";tag=as4d52b88e Call-ID: MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. CSeq: 1 INVITE Server: Asterisk PBX SVN-trunk-r180641 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4d3af189" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI.' in 6464 ms (Method: INVITE) LAsterisco*CLI> <--- SIP read from UDP:10.0.0.102:51858 ---> ACK sip:1999@10.0.2.7 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.102:51858;branch=z9hG4bK-d8754z-d61ce123ac58964d-1---d8754z-;rport To: "1999";tag=as4d52b88e From: "1012";tag=a7316a35 Call-ID: MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- LAsterisco*CLI> <--- SIP read from UDP:10.0.0.102:51858 ---> INVITE sip:1999@10.0.2.7 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.102:51858;branch=z9hG4bK-d8754z-9e07390bfa0e130a-1---d8754z-;rport Max-Forwards: 70 Contact: To: "1999" From: "1012";tag=a7316a35 Call-ID: MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="1012",realm="asterisk",nonce="4d3af189",uri="sip:1999@10.0.2.7",response="f98405af7541f67f8ba717215344070e",algorithm=MD5 Content-Length: 874 v=0 o=- 3 2 IN IP4 10.0.0.102 s=CounterPath X-Lite 3.0 c=IN IP4 10.0.0.102 t=0 0 m=audio 48486 RTP/AVP 107 119 100 106 0 105 98 8 3 101 a=alt:1 10 : kF2I1u+G Sq4/O0pW 10.0.0.102 48486 a=alt:2 9 : KWRNSyiG VvDtHmYr 192.168.1.69 48486 a=alt:3 8 : LBz3mYzX uZQc8jSV 192.168.2.166 48486 a=alt:4 7 : VZMcem8i /4VgK0TP 192.168.2.11 48486 a=alt:5 6 : +pMMAjkA qVmyVw4Y 192.168.0.169 48486 a=alt:6 5 : h3/G7t2Q 3wwi7QBK 192.168.8.164 48486 a=alt:7 4 : W6SmDyuj Uf0tNweB 169.254.1.58 48486 a=alt:8 3 : IyNxodns EXWVfJ7a 192.168.222.69 48486 a=alt:9 2 : qPAh2cuh BJW1ixVZ 192.168.136.1 48486 a=alt:10 1 : 3s9JLXyc OK2aOwkp 192.168.56.1 48486 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (13 headers 25 lines) --- Sending to 10.0.0.102 : 51858 (NAT) Using INVITE request as basis request - MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. Found peer '1012' for '1012' from 10.0.0.102:51858 Found RTP audio format 107 Found RTP audio format 119 Found RTP audio format 100 Found RTP audio format 106 Found RTP audio format 0 Found RTP audio format 105 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 10.0.0.102:48486 Found audio description format BV32 for ID 107 Found audio description format BV32-FEC for ID 119 Found audio description format SPEEX for ID 100 Found audio description format SPEEX-FEC for ID 106 Found audio description format SPEEX-FEC for ID 105 Found audio description format iLBC for ID 98 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc00040e (gsm|ulaw|alaw|ilbc|red|t140)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.0.102:48486 Looking for 1999 in deLocal (domain 10.0.2.7) list_route: hop: <--- Transmitting (no NAT) to 10.0.0.102:51858 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.102:51858;branch=z9hG4bK-d8754z-9e07390bfa0e130a-1---d8754z-;received=10.0.0.102;rport=51858 From: "1012";tag=a7316a35 To: "1999" Call-ID: MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r180641 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 10.0.2.7 port 27044 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.0.0.102:51858 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.102:51858;branch=z9hG4bK-d8754z-9e07390bfa0e130a-1---d8754z-;received=10.0.0.102;rport=51858 From: "1012";tag=a7316a35 To: "1999";tag=as6433bcdc Call-ID: MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r180641 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 308 v=0 o=root 59703134 59703134 IN IP4 10.0.2.7 s=Asterisk PBX SVN-trunk-r180641 c=IN IP4 10.0.2.7 t=0 0 m=audio 27044 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> Audio is at 10.0.2.7 port 27078 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.27:5062: INVITE sip:1001@10.0.0.27:5062 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK0b086df6;rport Max-Forwards: 70 From: "1012" ;tag=as2f531ccc To: Contact: Call-ID: 0a0b558406230ec17cee55a15b541d2a@10.0.2.7 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r180641 Date: Mon, 09 Mar 2009 14:31:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 310 v=0 o=root 913446364 913446364 IN IP4 10.0.2.7 s=Asterisk PBX SVN-trunk-r180641 c=IN IP4 10.0.2.7 t=0 0 m=audio 27078 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- LAsterisco*CLI> <--- SIP read from UDP:10.0.0.27:5062 ---> SIP/2.0 100 Trying To: From: "1012" ;tag=as2f531ccc Call-ID: 0a0b558406230ec17cee55a15b541d2a@10.0.2.7 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK0b086df6 Server: Linksys/SPA942-5.1.15(a) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- LAsterisco*CLI> <--- SIP read from UDP:10.0.0.27:5062 ---> SIP/2.0 180 Ringing To: ;tag=36f95fc296579053i2 From: "1012" ;tag=as2f531ccc Call-ID: 0a0b558406230ec17cee55a15b541d2a@10.0.2.7 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK0b086df6 Server: Linksys/SPA942-5.1.15(a) Content-Length: 0 Allow-Events: dialog <-------------> --- (9 headers 0 lines) --- LAsterisco*CLI> Scheduling destruction of SIP dialog '0a0b558406230ec17cee55a15b541d2a@10.0.2.7' in 6400 ms (Method: INVITE) Reliably Transmitting (no NAT) to 10.0.0.27:5062: CANCEL sip:1001@10.0.0.27:5062 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK0b086df6;rport Max-Forwards: 70 From: "1012" ;tag=as2f531ccc To: Call-ID: 0a0b558406230ec17cee55a15b541d2a@10.0.2.7 CSeq: 102 CANCEL User-Agent: Asterisk PBX SVN-trunk-r180641 Content-Length: 0 --- Scheduling destruction of SIP dialog '0a0b558406230ec17cee55a15b541d2a@10.0.2.7' in 6400 ms (Method: INVITE) LAsterisco*CLI> <--- SIP read from UDP:10.0.0.27:5062 ---> SIP/2.0 487 Request Terminated To: ;tag=36f95fc296579053i2 From: "1012" ;tag=as2f531ccc Call-ID: 0a0b558406230ec17cee55a15b541d2a@10.0.2.7 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK0b086df6 Server: Linksys/SPA942-5.1.15(a) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 10.0.0.27:5062: ACK sip:1001@10.0.0.27:5062 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK0b086df6;rport Max-Forwards: 70 From: "1012" ;tag=as2f531ccc To: ;tag=36f95fc296579053i2 Contact: Call-ID: 0a0b558406230ec17cee55a15b541d2a@10.0.2.7 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r180641 Content-Length: 0 --- LAsterisco*CLI> <--- SIP read from UDP:10.0.0.27:5062 ---> SIP/2.0 200 OK To: ;tag=36f95fc296579053i2 From: "1012" ;tag=as2f531ccc Call-ID: 0a0b558406230ec17cee55a15b541d2a@10.0.2.7 CSeq: 102 CANCEL Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK0b086df6 Server: Linksys/SPA942-5.1.15(a) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '0a0b558406230ec17cee55a15b541d2a@10.0.2.7' Method: INVITE LAsterisco*CLI> <--- SIP read from UDP:10.0.0.102:51858 ---> CANCEL sip:1999@10.0.2.7 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.102:51858;branch=z9hG4bK-d8754z-9e07390bfa0e130a-1---d8754z-;rport To: "1999" From: "1012";tag=a7316a35 Call-ID: MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. CSeq: 2 CANCEL User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="1012",realm="asterisk",nonce="4d3af189",uri="sip:1999@10.0.2.7",response="eb6c94aedf2496498b263b9bea953047",algorithm=MD5 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 10.0.0.102 : 51858 (NAT) <--- Reliably Transmitting (NAT) to 10.0.0.102:51858 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.0.0.102:51858;branch=z9hG4bK-d8754z-9e07390bfa0e130a-1---d8754z-;received=10.0.0.102;rport=51858 From: "1012";tag=a7316a35 To: "1999";tag=as6433bcdc Call-ID: MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. CSeq: 2 INVITE Server: Asterisk PBX SVN-trunk-r180641 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> <--- Transmitting (NAT) to 10.0.0.102:51858 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.102:51858;branch=z9hG4bK-d8754z-9e07390bfa0e130a-1---d8754z-;received=10.0.0.102;rport=51858 From: "1012";tag=a7316a35 To: "1999";tag=as6433bcdc Call-ID: MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. CSeq: 2 CANCEL Server: Asterisk PBX SVN-trunk-r180641 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> [Mar 9 15:32:01] WARNING[8915]: channel.c:3156 ast_prod: Prodding channel 'SIP/1012-08416a48' failed Audio is at 10.0.2.7 port 29492 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.27:5062: INVITE sip:1001@10.0.0.27:5062 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK5c41a8b8;rport Max-Forwards: 70 From: "1012" ;tag=as563b2401 To: Contact: Call-ID: 3578fe3f0ca981ea193762b3206d4e95@10.0.2.7 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r180641 Date: Mon, 09 Mar 2009 14:32:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 310 v=0 o=root 319828340 319828340 IN IP4 10.0.2.7 s=Asterisk PBX SVN-trunk-r180641 c=IN IP4 10.0.2.7 t=0 0 m=audio 29492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Scheduling destruction of SIP dialog '3578fe3f0ca981ea193762b3206d4e95@10.0.2.7' in 6400 ms (Method: INVITE) <--- SIP read from UDP:10.0.0.102:51858 ---> ACK sip:1999@10.0.2.7 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.102:51858;branch=z9hG4bK-d8754z-9e07390bfa0e130a-1---d8754z-;rport To: "1999";tag=as6433bcdc From: "1012";tag=a7316a35 Call-ID: MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI. CSeq: 2 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog 'MDE0NzYzNTJmOTg4Mzg3ZTIyYThkZGI5ZDkxYjNmYjI.' Method: ACK LAsterisco*CLI> <--- SIP read from UDP:10.0.0.27:5062 ---> SIP/2.0 100 Trying To: From: "1012" ;tag=as563b2401 Call-ID: 3578fe3f0ca981ea193762b3206d4e95@10.0.2.7 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK5c41a8b8 Server: Linksys/SPA942-5.1.15(a) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Reliably Transmitting (no NAT) to 10.0.0.27:5062: CANCEL sip:1001@10.0.0.27:5062 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK5c41a8b8;rport Max-Forwards: 70 From: "1012" ;tag=as563b2401 To: Call-ID: 3578fe3f0ca981ea193762b3206d4e95@10.0.2.7 CSeq: 102 CANCEL User-Agent: Asterisk PBX SVN-trunk-r180641 Content-Length: 0 --- Scheduling destruction of SIP dialog '3578fe3f0ca981ea193762b3206d4e95@10.0.2.7' in 6400 ms (Method: INVITE) LAsterisco*CLI> <--- SIP read from UDP:10.0.0.27:5062 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist To: ;tag=1f2cf5f9e89b93e6i2 From: "1012" ;tag=as563b2401 Call-ID: 3578fe3f0ca981ea193762b3206d4e95@10.0.2.7 CSeq: 102 CANCEL Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK5c41a8b8 Server: Linksys/SPA942-5.1.15(a) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Mar 9 15:32:01] WARNING[8807]: chan_sip.c:17436 handle_response: Remote host can't match request CANCEL to call '3578fe3f0ca981ea193762b3206d4e95@10.0.2.7'. Giving up. LAsterisco*CLI> <--- SIP read from UDP:10.0.0.27:5062 ---> SIP/2.0 180 Ringing To: ;tag=18639c71efa7a67i2 From: "1012" ;tag=as563b2401 Call-ID: 3578fe3f0ca981ea193762b3206d4e95@10.0.2.7 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.2.7:5060;branch=z9hG4bK5c41a8b8 Server: Linksys/SPA942-5.1.15(a) Content-Length: 0 Allow-Events: dialog <-------------> --- (9 headers 0 lines) --- LAsterisco*CLI> Disconnected from Asterisk server