Asterisk SVN-trunk-r69583, Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer *CLI> sip set debug on SIP Debugging enabled *CLI> <--- SIP read from 132.64.4.137:2051 ---> SUBSCRIBE sip:80620@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-sqyp1abj0b80;rport From: ;tag=lg9fclqq1m To: Call-ID: 3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2051 (NAT) Found peer '80602' for '80602' from 132.64.4.137:2051 <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-sqyp1abj0b80;received=132.64.4.137;rport=2051 From: ;tag=lg9fclqq1m To: ;tag=as7f5a1301 Call-ID: 3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="4b9cbf7c" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from 132.64.4.137:2051 ---> INVITE sip:80620@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-hwzgus06pq01;rport From: ;tag=gtc8ard2ci To: Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 475 v=0 o=root 2102981971 2102981971 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 51068 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:/lXmT/cTtRL5Zbc7N7wu4cpjgSfpGdGsl5vT16mz a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (18 headers 19 lines) ---  == Using TOS bits 0  == Using CoS mark 5  == Using TOS bits 0  == Using CoS mark 6 Sending to 132.64.4.137 : 2051 (NAT) Using INVITE request as basis request - 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 Found user '80602' for '80602' <--- Reliably Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-hwzgus06pq01;received=132.64.4.137;rport=2051 From: ;tag=gtc8ard2ci To: ;tag=as3881e997 Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="55d2818a" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4' in 32000 ms (Method: INVITE) <--- SIP read from 132.64.4.137:2051 ---> SUBSCRIBE sip:80620@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-wxz29nkw4iko;rport From: ;tag=lg9fclqq1m To: Call-ID: 3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4 CSeq: 2 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Authorization: Digest username="80602",realm="cc.huji.ac.il",nonce="4b9cbf7c",uri="sip:80620@pbx-dev.cc.huji.ac.il;user=phone",response="2aa696d575bdb6bc8b55f78ebc85e9cd",algorithm=md5 Expires: 3600 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2051 (NAT) Found peer '80602' for '80602' from 132.64.4.137:2051 Looking for 80620 in huji-remote-gr (domain pbx-dev.cc.huji.ac.il) Scheduling destruction of SIP dialog '3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4' in 3610000 ms (Method: SUBSCRIBE) [Jun 17 08:15:22] NOTICE[28764]: chan_sip.c:15375 handle_request_subscribe: Got SUBSCRIBE for extension 80620@huji-remote-gr from 132.64.4.137, but there is no hint for that extension. <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-wxz29nkw4iko;received=132.64.4.137;rport=2051 From: ;tag=lg9fclqq1m To: ;tag=as7f5a1301 Call-ID: 3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Really destroying SIP dialog '3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4' Method: SUBSCRIBE <--- SIP read from 132.64.4.137:2051 ---> ACK sip:80620@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-hwzgus06pq01;rport From: ;tag=gtc8ard2ci To: ;tag=as3881e997 Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 132.64.4.137:2051 ---> INVITE sip:80620@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-kv163z7zindu;rport From: ;tag=gtc8ard2ci To: Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="80602",realm="cc.huji.ac.il",nonce="55d2818a",uri="sip:80620@pbx-dev.cc.huji.ac.il;user=phone",response="4d0ecbf0a1825c112245fccd434da62b",algorithm=md5 Content-Type: application/sdp Content-Length: 475 v=0 o=root 2102981971 2102981971 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 51068 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:/lXmT/cTtRL5Zbc7N7wu4cpjgSfpGdGsl5vT16mz a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (19 headers 19 lines) --- Sending to 132.64.4.137 : 2051 (NAT) Using INVITE request as basis request - 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 Found user '80602' for '80602' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.137:51068 Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format g722 for ID 9 Found description format g726-32 for ID 2 Found description format gsm for ID 3 Found description format g729 for ID 18 Found description format g723 for ID 4 Found description format telephone-event for ID 101 Capabilities: us - 0xc0008 (alaw|h261|h263), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:51068 Looking for 80620 in huji-remote-gr (domain pbx-dev.cc.huji.ac.il) list_route: hop: <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-kv163z7zindu;received=132.64.4.137;rport=2051 From: ;tag=gtc8ard2ci To: Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------>  -- Executing [80620@huji-remote-gr:1] Set("SIP/80602-0985bd38", "_To=80620") in new stack  -- Executing [80620@huji-remote-gr:2] Set("SIP/80602-0985bd38", "_From=80602") in new stack  -- Executing [80620@huji-remote-gr:3] Set("SIP/80602-0985bd38", "DB(80620/LastCaller)=80602") in new stack  -- Executing [80620@huji-remote-gr:4] Set("SIP/80602-0985bd38", "DB(80602/LastCalled)=80620") in new stack  -- Executing [80620@huji-remote-gr:5] MYSQL("SIP/80602-0985bd38", "Connect connid localhost asterisk NotMe43e080 asterisk") in new stack  -- Executing [80620@huji-remote-gr:6] MYSQL("SIP/80602-0985bd38", "Query resID 1 SELECT additional_numbers from shared_lines where orig_number=80620") in new stack  -- Executing [80620@huji-remote-gr:7] MYSQL("SIP/80602-0985bd38", "Fetch FetchId 2 aEXTEN") in new stack  -- Executing [80620@huji-remote-gr:8] NoOp("SIP/80602-0985bd38", "") in new stack  -- Executing [80620@huji-remote-gr:9] MYSQL("SIP/80602-0985bd38", "Clear 2") in new stack  -- Executing [80620@huji-remote-gr:10] MYSQL("SIP/80602-0985bd38", "Query resID 1 SELECT callerid from sip_users where name=80620") in new stack  -- Executing [80620@huji-remote-gr:11] MYSQL("SIP/80602-0985bd38", "Fetch FetchId 2 CalledName") in new stack  -- Executing [80620@huji-remote-gr:12] NoOp("SIP/80602-0985bd38", "Yehavi Bourvine <80620>") in new stack  -- Executing [80620@huji-remote-gr:13] MYSQL("SIP/80602-0985bd38", "Clear 2") in new stack  -- Executing [80620@huji-remote-gr:14] MYSQL("SIP/80602-0985bd38", "Disconnect 1") in new stack  -- Executing [80620@huji-remote-gr:15] NoOp("SIP/80602-0985bd38", "") in new stack  -- Executing [80620@huji-remote-gr:16] Set("SIP/80602-0985bd38", "Status=NOT_INUSE") in new stack  -- Executing [80620@huji-remote-gr:17] NoOp("SIP/80602-0985bd38", "NOT_INUSE") in new stack  -- Executing [80620@huji-remote-gr:18] GotoIf("SIP/80602-0985bd38", "1?OK:WAITING_CALL") in new stack  -- Goto (huji-remote-gr,80620,20)  -- Executing [80620@huji-remote-gr:20] Dial("SIP/80602-0985bd38", "SIP/80620|20|") in new stack  == Using TOS bits 0  == Using CoS mark 5  == Using TOS bits 0  == Using CoS mark 6 Audio is at 132.64.9.164 port 11646 Video is at 132.64.9.164 port 10122 Adding codec 0x8 (alaw) to SDP Adding video codec 0x40000 (h261) to SDP Adding video codec 0x80000 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.156:5060: INVITE sip:80620@132.64.4.156:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK44064a8d;rport Max-Forwards: 70 From: "Naama Nahmias" ;tag=as320df247 To: Contact: Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r69583 Remote-Party-ID: "Naama Nahmias" ;privacy=off;screen=no Date: Sun, 17 Jun 2007 05:15:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 347 v=0 o=root 760379830 760379830 IN IP4 132.64.9.164 s=session c=IN IP4 132.64.9.164 b=CT:384 t=0 0 m=audio 11646 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 10122 RTP/AVP 31 34 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=sendrecv ---  -- Called 80620 <--- SIP read from 132.64.4.156:5060 ---> SIP/2.0 100 Trying Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 102 INVITE From: "Naama Nahmias" ;tag=as320df247 To: ;tag=ec9bef3b8b97506 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK44064a8d;rport Content-Length: 0 User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 132.64.4.156:5060 ---> SIP/2.0 180 Ringing Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 102 INVITE From: "Naama Nahmias" ;tag=as320df247 To: ;tag=ec9bef3b8b97506 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK44064a8d;rport Content-Length: 0 Contact: User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> --- (9 headers 0 lines) ---  -- SIP/80620-09856df0 is ringing <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-kv163z7zindu;received=132.64.4.137;rport=2051 From: ;tag=gtc8ard2ci To: ;tag=as33df891d Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from 132.64.4.156:5060 ---> SIP/2.0 200 OK Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 102 INVITE From: "Naama Nahmias" ;tag=as320df247 To: ;tag=ec9bef3b8b97506 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK44064a8d;rport Content-Length: 223 Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Supported: replaces Contact: User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 393107995 IN IP4 132.64.4.156 s=SIP Call c=IN IP4 132.64.4.156 t=0 0 m=audio 34008 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv m=video 0 RTP/AVP 31 34 <-------------> --- (18 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found RTP video format 31 Found RTP video format 34 Peer audio RTP is at port 132.64.4.156:34008 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0xc0008 (alaw|h261|h263), peer - audio=0xc0008 (alaw|h261|h263)/video=0xc0000 (h261|h263)/text=0x0 (nothing), combined - 0xc0008 (alaw|h261|h263) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.156:34008 --- set_address_from_contact host '132.64.4.156' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.156, port 5060 Transmitting (no NAT) to 132.64.4.156:5060: ACK sip:80620@132.64.4.156:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK1a64461e;rport Max-Forwards: 70 From: "Naama Nahmias" ;tag=as320df247 To: ;tag=ec9bef3b8b97506 Contact: Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r69583 Remote-Party-ID: "Naama Nahmias" ;privacy=off;screen=no Content-Length: 0 ---  -- SIP/80620-09856df0 answered SIP/80602-0985bd38 Audio is at 132.64.9.164 port 16318 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-kv163z7zindu;received=132.64.4.137;rport=2051 From: ;tag=gtc8ard2ci To: ;tag=as33df891d Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 250 v=0 o=root 1188567553 1188567553 IN IP4 132.64.9.164 s=session c=IN IP4 132.64.9.164 t=0 0 m=audio 16318 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 132.64.4.137:2051 ---> ACK sip:80620@132.64.9.164 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-33ho4dk2w6b4;rport From: ;tag=gtc8ard2ci To: ;tag=as33df891d Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 132.64.4.137:2051 ---> SUBSCRIBE sip:80620@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-rr8tbwxqh3tg;rport From: ;tag=lg9fclqq1m To: Call-ID: 3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4 CSeq: 3 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Expires: 0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2051 (NAT) Found peer '80602' for '80602' from 132.64.4.137:2051 <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-rr8tbwxqh3tg;received=132.64.4.137;rport=2051 From: ;tag=lg9fclqq1m To: ;tag=as0c00f5bd Call-ID: 3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4 CSeq: 3 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="6943000a" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from 132.64.4.137:2051 ---> SUBSCRIBE sip:80620@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-ci1qc8dgp142;rport From: ;tag=lg9fclqq1m To: Call-ID: 3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4 CSeq: 4 SUBSCRIBE Max-Forwards: 70 Contact: ;flow-id=1 Event: dialog;purpose=call-completion Accept: application/dialog-info+xml Authorization: Digest username="80602",realm="cc.huji.ac.il",nonce="6943000a",uri="sip:80620@pbx-dev.cc.huji.ac.il;user=phone",response="1e3ee9717b3853ddbd6328692aeb254e",algorithm=md5 Expires: 0 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 132.64.4.137 : 2051 (NAT) Found peer '80602' for '80602' from 132.64.4.137:2051 Looking for 80620 in huji-remote-gr (domain pbx-dev.cc.huji.ac.il) [Jun 17 08:15:26] NOTICE[28764]: chan_sip.c:15375 handle_request_subscribe: Got SUBSCRIBE for extension 80620@huji-remote-gr from 132.64.4.137, but there is no hint for that extension. <--- Transmitting (no NAT) to 132.64.4.137:2051 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 132.64.4.137:2051;branch=z9hG4bK-ci1qc8dgp142;received=132.64.4.137;rport=2051 From: ;tag=lg9fclqq1m To: ;tag=as0c00f5bd Call-ID: 3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4 CSeq: 4 SUBSCRIBE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Really destroying SIP dialog '3c2683d20ea6-0pfwjtayy4e3@snom320-0004132480B4' Method: SUBSCRIBE  -- Native bridging SIP/80602-0985bd38 and SIP/80620-09856df0 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Audio is at 132.64.9.164 port 16318 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.137:2051: INVITE sip:80602@132.64.4.137:2051;line=ydg4eu9f SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK02af7228;rport Max-Forwards: 70 From: ;tag=as33df891d To: ;tag=gtc8ard2ci Contact: Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 250 v=0 o=root 1188567553 1188567554 IN IP4 132.64.4.156 s=session c=IN IP4 132.64.4.156 t=0 0 m=audio 34008 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.156, port 5060 Audio is at 132.64.9.164 port 11646 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.156:5060: INVITE sip:80620@132.64.4.156:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK7a9e4b58;rport Max-Forwards: 70 From: "Naama Nahmias" ;tag=as320df247 To: ;tag=ec9bef3b8b97506 Contact: Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-trunk-r69583 Remote-Party-ID: "Naama Nahmias" ;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 248 v=0 o=root 760379830 760379831 IN IP4 132.64.4.137 s=session c=IN IP4 132.64.4.137 t=0 0 m=audio 51068 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 132.64.4.137:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK02af7228;rport=5060 From: ;tag=as33df891d To: ;tag=gtc8ard2ci Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 327 v=0 o=root 2102981971 2102981972 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 51068 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:/lXmT/cTtRL5Zbc7N7wu4cpjgSfpGdGsl5vT16mz a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (13 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.137:51068 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0xc0008 (alaw|h261|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:51068 --- set_address_from_contact host '132.64.4.137' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Transmitting (no NAT) to 132.64.4.137:2051: ACK sip:80602@132.64.4.137:2051;line=ydg4eu9f SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK356c89cf;rport Max-Forwards: 70 From: ;tag=as33df891d To: ;tag=gtc8ard2ci Contact: Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r69583 Content-Length: 0 --- <--- SIP read from 132.64.4.156:5060 ---> SIP/2.0 100 Trying Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 103 INVITE From: "Naama Nahmias" ;tag=as320df247 To: ;tag=ec9bef3b8b97506 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK7a9e4b58;rport Content-Length: 0 User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 132.64.4.156:5060 ---> SIP/2.0 200 OK Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 103 INVITE From: "Naama Nahmias" ;tag=as320df247 To: ;tag=ec9bef3b8b97506 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK7a9e4b58;rport Content-Length: 198 Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Supported: replaces Contact: User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 393107995 IN IP4 132.64.4.156 s=SIP Call c=IN IP4 132.64.4.156 t=0 0 m=audio 34008 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv <-------------> --- (18 headers 10 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.156:34008 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.156:34008 --- set_address_from_contact host '132.64.4.156' set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.156, port 5060 Transmitting (no NAT) to 132.64.4.156:5060: ACK sip:80620@132.64.4.156:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK0b7bb590;rport Max-Forwards: 70 From: "Naama Nahmias" ;tag=as320df247 To: ;tag=ec9bef3b8b97506 Contact: Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-trunk-r69583 Remote-Party-ID: "Naama Nahmias" ;privacy=off;screen=no Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Audio is at 132.64.9.164 port 16318 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.137:2051: INVITE sip:80602@132.64.4.137:2051;line=ydg4eu9f SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK442c9805;rport Max-Forwards: 70 From: ;tag=as33df891d To: ;tag=gtc8ard2ci Contact: Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 250 v=0 o=root 1188567553 1188567555 IN IP4 132.64.4.156 s=session c=IN IP4 132.64.4.156 t=0 0 m=audio 34008 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Internal RTCP NTP clock skew detected: lsr=842022961, now=1106183244, dlsr=858926646 (13106:180ms), diff=594766363 <--- SIP read from 132.64.4.137:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK442c9805;rport=5060 From: ;tag=as33df891d To: ;tag=gtc8ard2ci Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 103 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 327 v=0 o=root 2102981971 2102981973 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 51068 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:/lXmT/cTtRL5Zbc7N7wu4cpjgSfpGdGsl5vT16mz a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (13 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 132.64.4.137:51068 Found description format pcma for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:51068 --- set_address_from_contact host '132.64.4.137' set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Transmitting (no NAT) to 132.64.4.137:2051: ACK sip:80602@132.64.4.137:2051;line=ydg4eu9f SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK7a345c33;rport Max-Forwards: 70 From: ;tag=as33df891d To: ;tag=gtc8ard2ci Contact: Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-trunk-r69583 Content-Length: 0 --- Here comes the HOLD event <======================= *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> <--- SIP read from 132.64.4.156:5060 ---> INVITE sip:80602@132.64.9.164 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.156:5060;branch=z9hG4bK042904057 Max-Forwards: 70 Content-Length: 258 To: "Naama Nahmias" ;tag=as320df247 From: ;tag=ec9bef3b8b97506 Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 99405417 INVITE Supported: timer Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Contact: Supported: replaces User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 393107996 IN IP4 132.64.4.156 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 34008 RTP/AVP 0 8 18 2 127 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:127 telephone-event/8000 a=ptime:20 <-------------> --- (20 headers 12 lines) --- Sending to 132.64.4.156 : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 127 Peer audio RTP is at port 0.0.0.0:34008 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format G726-32 for ID 2 Found description format telephone-event for ID 127 Capabilities: us - 0x8 (alaw), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 0.0.0.0:34008 Audio is at 132.64.9.164 port 11646 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 132.64.4.156:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.156:5060;branch=z9hG4bK042904057;received=132.64.4.156 From: ;tag=ec9bef3b8b97506 To: "Naama Nahmias" ;tag=as320df247 Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 99405417 INVITE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 248 v=0 o=root 760379830 760379832 IN IP4 132.64.4.137 s=session c=IN IP4 132.64.4.137 t=0 0 m=audio 51068 RTP/AVP 8 127 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Audio is at 132.64.9.164 port 16318 Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.137:2051: INVITE sip:80602@132.64.4.137:2051;line=ydg4eu9f SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK63c1158a;rport Max-Forwards: 70 From: ;tag=as33df891d To: ;tag=gtc8ard2ci Contact: Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 104 INVITE User-Agent: Asterisk PBX SVN-trunk-r69583 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 214 v=0 o=root 1188567553 1188567556 IN IP4 132.64.9.164 s=session c=IN IP4 132.64.9.164 t=0 0 m=audio 16318 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv ---  -- Started music on hold, class 'default', on SIP/80602-0985bd38 <--- SIP read from 132.64.4.137:2051 ---> SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK63c1158a;rport=5060 From: ;tag=as33df891d To: ;tag=gtc8ard2ci Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 104 INVITE Contact: ;flow-id=1 Warning: 304 x-snom-adr "No supported media type found" Content-Length: 0 <-------------> --- (9 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Transmitting (no NAT) to 132.64.4.137:2051: ACK sip:80602@132.64.4.137:2051;line=ydg4eu9f SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK63c1158a;rport Max-Forwards: 70 From: ;tag=as33df891d To: ;tag=gtc8ard2ci Contact: Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 104 ACK User-Agent: Asterisk PBX SVN-trunk-r69583 Content-Length: 0 --- Scheduling destruction of SIP dialog '29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164' in 32000 ms (Method: INVITE)  == Spawn extension (huji-remote-gr, 80620, 20) exited non-zero on 'SIP/80602-0985bd38'  -- Executing [h@huji-remote-gr:1] NoOp("SIP/80602-0985bd38", "80602 80620 h") in new stack  -- Executing [h@huji-remote-gr:2] Set("SIP/80602-0985bd38", "tmp=") in new stack  -- Executing [h@huji-remote-gr:3] NoOp("SIP/80602-0985bd38", "80602 ") in new stack  -- Executing [h@huji-remote-gr:4] GotoIf("SIP/80602-0985bd38", "?5:103") in new stack  -- Goto (huji-remote-gr,h,103)  -- Executing [h@huji-remote-gr:103] NoOp("SIP/80602-0985bd38", "Nothing to call") in new stack  -- Stopped music on hold on SIP/80602-0985bd38 Scheduling destruction of SIP dialog '3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2051 Reliably Transmitting (no NAT) to 132.64.4.137:2051: BYE sip:80602@132.64.4.137:2051;line=ydg4eu9f SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK6f18b5d3;rport Max-Forwards: 70 From: ;tag=as33df891d To: ;tag=gtc8ard2ci Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 105 BYE User-Agent: Asterisk PBX SVN-trunk-r69583 Content-Length: 0 --- <--- SIP read from 132.64.4.156:5060 ---> ACK sip:80602@132.64.9.164 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.156:5060;branch=z9hG4bKef128e4c2 Max-Forwards: 70 Content-Length: 0 To: "Naama Nahmias" ;tag=as320df247 From: ;tag=ec9bef3b8b97506 Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 99405417 ACK Contact: User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.156, port 5060 Reliably Transmitting (no NAT) to 132.64.4.156:5060: BYE sip:80620@132.64.4.156:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK28dfc7ce;rport Max-Forwards: 70 From: "Naama Nahmias" ;tag=as320df247 To: ;tag=ec9bef3b8b97506 Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 104 BYE User-Agent: Asterisk PBX SVN-trunk-r69583 Remote-Party-ID: "Naama Nahmias" ;privacy=off;screen=no Content-Length: 0 --- Scheduling destruction of SIP dialog '29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164' in 32000 ms (Method: ACK) <--- SIP read from 132.64.4.156:5060 ---> SIP/2.0 200 OK Call-ID: 29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164 CSeq: 104 BYE From: "Naama Nahmias" ;tag=as320df247 To: ;tag=ec9bef3b8b97506 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK28dfc7ce;rport Content-Length: 0 Supported: replaces User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '29ac5db966beb3ca5028f9cd1794a65d@132.64.9.164' Method: ACK <--- SIP read from 132.64.4.137:2051 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK6f18b5d3;rport=5060 From: ;tag=as33df891d To: ;tag=gtc8ard2ci Call-ID: 3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4 CSeq: 105 BYE Contact: ;flow-id=1 User-Agent: snom320/6.2.2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '3c2683d1e7ef-yg3i5scbs2xq@snom320-0004132480B4' Method: ACK *CLI>