[astadmin@nytadla01 ~]$ asterisk -r Asterisk 1.8.7.1, Copyright (C) 1999 - 2011 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 1.8.7.1 currently running on nytadla01 (pid = 23307) Verbosity is at least 9 Core debug is at least 9 == Manager 'ChatterBox' logged off from 10.221.196.232 == Manager 'ChatterBox' logged on from 10.221.196.232 -- Executing [18366@callback:1] Log("Local/18366@callback-090a;2", "notice,Channel:Local/18366@callback-090a;2: Callback request on:18366") in new stack [Dec 2 02:33:56] NOTICE[28863]: Ext. 18366:1 @ callback: Channel:Local/18366@callback-090a;2: Callback request on:18366 -- Executing [18366@callback:2] Goto("Local/18366@callback-090a;2", "callback-primary,18366,1") in new stack -- Goto (callback-primary,18366,1) -- Executing [18366@callback-primary:1] Set("Local/18366@callback-090a;2", "CALLERID(all)=Naidu, Abhishek<18366>") in new stack -- Executing [18366@callback-primary:2] Macro("Local/18366@callback-090a;2", "callback,SIP/sip_trunk_primary/18366,callback-secondary,18366,true") in new stack -- Executing [s@macro-callback:1] Log("Local/18366@callback-090a;2", "notice,Channel:Local/18366@callback-090a;2: Callback macro dialling SIP/sip_trunk_primary/18366...") in new stack [Dec 2 02:33:56] NOTICE[28863]: Ext. s:1 @ macro-callback: Channel:Local/18366@callback-090a;2: Callback macro dialling SIP/sip_trunk_primary/18366... -- Executing [s@macro-callback:2] Dial("Local/18366@callback-090a;2", "SIP/sip_trunk_primary/18366,30") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.25.18.188:5060: INVITE sip:18366@172.25.18.188:5060 SIP/2.0 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK2cff5314 Max-Forwards: 70 From: "Naidu, Abhishek" ;tag=as271f7c90 To: Contact: Call-ID: 1e9d7cbc674045625fada08b3000cc1a@138.8.237.64:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.7.1 Date: Fri, 02 Dec 2011 07:33:56 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 280 v=0 o=root 803500342 803500342 IN IP4 138.8.237.64 s=GS Asterisk PBX c=IN IP4 138.8.237.64 t=0 0 m=audio 21904 RTP/AVP 0 8 101 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 --- -- Called SIP/sip_trunk_primary/18366 <--- SIP read from UDP:172.25.18.188:5060 ---> SIP/2.0 100 Trying Date: Fri, 02 Dec 2011 07:34:11 GMT From: "Naidu, Abhishek" ;tag=as271f7c90 Allow-Events: presence Content-Length: 0 To: Call-ID: 1e9d7cbc674045625fada08b3000cc1a@138.8.237.64:5060 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK2cff5314 CSeq: 102 INVITE <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:172.25.18.188:5060 ---> SIP/2.0 180 Ringing Date: Fri, 02 Dec 2011 07:34:11 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "Naidu, Abhishek" ;tag=as271f7c90 Allow-Events: presence P-Asserted-Identity: "test nortel mwi" Supported: X-cisco-srtp-fallback Supported: Geolocation Remote-Party-ID: "test nortel mwi" ;party=called;screen=yes;privacy=off Content-Length: 0 To: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753173 Contact: Call-ID: 1e9d7cbc674045625fada08b3000cc1a@138.8.237.64:5060 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK2cff5314 CSeq: 102 INVITE <-------------> --- (15 headers 0 lines) --- -- SIP/sip_trunk_primary-00000026 is ringing <--- SIP read from UDP:172.25.18.188:5060 ---> SIP/2.0 200 OK Date: Fri, 02 Dec 2011 07:34:11 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "Naidu, Abhishek" ;tag=as271f7c90 Allow-Events: presence, kpml P-Asserted-Identity: "Rodriguez, Timothy 18366" Supported: replaces Supported: X-cisco-srtp-fallback Supported: Geolocation Remote-Party-ID: "Rodriguez, Timothy 18366" ;party=called;screen=yes;privacy=off Content-Length: 213 Require: timer To: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753173 Contact: Content-Type: application/sdp Call-ID: 1e9d7cbc674045625fada08b3000cc1a@138.8.237.64:5060 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK2cff5314 CSeq: 102 INVITE Session-Expires: 1800;refresher=uas v=0 o=CiscoSystemsCCM-SIP 2000 1 IN IP4 172.25.18.188 s=SIP Call c=IN IP4 172.25.26.82 t=0 0 m=audio 28136 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (19 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 172.25.26.82:28136 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 172.25.18.188:5060 Transmitting (no NAT) to 172.25.18.188:5060: ACK sip:18366@172.25.18.188:5060 SIP/2.0 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK1adcbe4a Max-Forwards: 70 From: "Naidu, Abhishek" ;tag=as271f7c90 To: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753173 Contact: Call-ID: 1e9d7cbc674045625fada08b3000cc1a@138.8.237.64:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.7.1 Content-Length: 0 --- -- SIP/sip_trunk_primary-00000026 answered Local/18366@callback-090a;2 > Channel Local/18366@callback-090a;1 was answered. -- Executing [89991000@outgoing:1] Log("Local/18366@callback-090a;1", "notice,Channel:Local/18366@callback-090a;1: Callto request on:89991000") in new stack [Dec 2 02:33:58] NOTICE[28864]: Ext. 89991000:1 @ outgoing: Channel:Local/18366@callback-090a;1: Callto request on:89991000 -- Executing [89991000@outgoing:2] Goto("Local/18366@callback-090a;1", "callto-primary,89991000,1") in new stack -- Goto (callto-primary,89991000,1) -- Executing [89991000@callto-primary:1] Set("Local/18366@callback-090a;1", "CALLERID(all)=Naidu, Abhishek<18366>") in new stack -- Executing [89991000@callto-primary:2] Macro("Local/18366@callback-090a;1", "trunkdial,SIP/sip_trunk_primary/89991000,callto-secondary,89991000,true") in new stack -- Executing [s@macro-trunkdial:1] Log("Local/18366@callback-090a;1", "notice,Channel:Local/18366@callback-090a;1: Callto macro dialling SIP/sip_trunk_primary/89991000...") in new stack [Dec 2 02:33:58] NOTICE[28864]: Ext. s:1 @ macro-trunkdial: Channel:Local/18366@callback-090a;1: Callto macro dialling SIP/sip_trunk_primary/89991000... -- Executing [s@macro-trunkdial:2] Dial("Local/18366@callback-090a;1", "SIP/sip_trunk_primary/89991000,60") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.25.18.188:5060: INVITE sip:89991000@172.25.18.188:5060 SIP/2.0 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK6588c851 Max-Forwards: 70 From: "Naidu, Abhishek" ;tag=as29868285 To: Contact: Call-ID: 724975d27232fc186ed7a7c80a34adbc@138.8.237.64:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.7.1 Date: Fri, 02 Dec 2011 07:33:58 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 282 v=0 o=root 1403505908 1403505908 IN IP4 138.8.237.64 s=GS Asterisk PBX c=IN IP4 138.8.237.64 t=0 0 m=audio 30328 RTP/AVP 0 8 101 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 --- -- Called SIP/sip_trunk_primary/89991000 -- Local/18366@callback-090a;1 requested special control 20, passing it to SIP/sip_trunk_primary-00000027 <--- SIP read from UDP:172.25.18.188:5060 ---> SIP/2.0 100 Trying Date: Fri, 02 Dec 2011 07:34:13 GMT From: "Naidu, Abhishek" ;tag=as29868285 Allow-Events: presence Content-Length: 0 To: Call-ID: 724975d27232fc186ed7a7c80a34adbc@138.8.237.64:5060 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK6588c851 CSeq: 102 INVITE <-------------> --- (9 headers 0 lines) --- == Spawn extension (macro-callback, s, 2) exited non-zero on 'Local/18366@callback-090a;2' in macro 'callback' == Spawn extension (callback-primary, 18366, 2) exited non-zero on 'Local/18366@callback-090a;2' -- SIP/sip_trunk_primary-00000026 requested special control 20, passing it to SIP/sip_trunk_primary-00000027 <--- SIP read from UDP:172.25.18.188:5060 ---> SIP/2.0 180 Ringing Date: Fri, 02 Dec 2011 07:34:13 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "Naidu, Abhishek" ;tag=as29868285 Allow-Events: presence P-Asserted-Identity: Supported: X-cisco-srtp-fallback Supported: Geolocation Remote-Party-ID: ;party=called;screen=yes;privacy=off Content-Length: 0 To: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753175 Contact: Call-ID: 724975d27232fc186ed7a7c80a34adbc@138.8.237.64:5060 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK6588c851 CSeq: 102 INVITE <-------------> --- (15 headers 0 lines) --- -- SIP/sip_trunk_primary-00000027 is ringing <--- SIP read from UDP:172.25.18.188:5060 ---> SIP/2.0 183 Session Progress Date: Fri, 02 Dec 2011 07:34:13 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "Naidu, Abhishek" ;tag=as29868285 Allow-Events: presence, kpml P-Asserted-Identity: Supported: X-cisco-srtp-fallback Supported: Geolocation Remote-Party-ID: ;party=called;screen=yes;privacy=off Content-Length: 223 To: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753175 Contact: Content-Type: application/sdp Call-ID: 724975d27232fc186ed7a7c80a34adbc@138.8.237.64:5060 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK6588c851 CSeq: 102 INVITE v=0 o=CiscoSystemsCCM-SIP 2000 1 IN IP4 172.25.18.188 s=SIP Call c=IN IP4 172.25.19.4 t=0 0 m=audio 4000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendonly a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (16 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 172.25.19.4:4000 -- Call on SIP/sip_trunk_primary-00000027 placed on hold -- Music class default requested but no musiconhold loaded. -- SIP/sip_trunk_primary-00000027 is making progress passing it to SIP/sip_trunk_primary-00000026 <--- SIP read from UDP:172.25.18.188:5060 ---> SIP/2.0 200 OK Date: Fri, 02 Dec 2011 07:34:13 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "Naidu, Abhishek" ;tag=as29868285 Allow-Events: presence, kpml Supported: replaces Supported: X-cisco-srtp-fallback Supported: Geolocation Remote-Party-ID: ;party=called;screen=no;privacy=off Content-Length: 223 Require: timer To: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753175 Contact: Content-Type: application/sdp Call-ID: 724975d27232fc186ed7a7c80a34adbc@138.8.237.64:5060 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK6588c851 CSeq: 102 INVITE P-Preferred-Identity: Session-Expires: 1800;refresher=uas v=0 o=CiscoSystemsCCM-SIP 2000 1 IN IP4 172.25.18.188 s=SIP Call c=IN IP4 172.25.19.4 t=0 0 m=audio 4000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendonly a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (19 headers 11 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 172.25.18.188:5060 Transmitting (no NAT) to 172.25.18.188:5060: ACK sip:89991000@172.25.18.188:5060 SIP/2.0 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK1c59bd73 Max-Forwards: 70 From: "Naidu, Abhishek" ;tag=as29868285 To: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753175 Contact: Call-ID: 724975d27232fc186ed7a7c80a34adbc@138.8.237.64:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.7.1 Content-Length: 0 --- -- SIP/sip_trunk_primary-00000027 answered SIP/sip_trunk_primary-00000026 -- Locally bridging SIP/sip_trunk_primary-00000026 and SIP/sip_trunk_primary-00000027 <--- SIP read from UDP:172.25.18.188:5060 ---> INVITE sip:18366@138.8.237.64:5060 SIP/2.0 Date: Fri, 02 Dec 2011 07:34:15 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753175 Allow-Events: presence, kpml Supported: timer,resource-priority,replaces Min-SE: 1800 Remote-Party-ID: ;party=calling;screen=no;privacy=off Content-Length: 212 User-Agent: Cisco-CUCM7.1 To: "Naidu, Abhishek" ;tag=as29868285 Contact: Expires: 180 Content-Type: application/sdp Call-ID: 724975d27232fc186ed7a7c80a34adbc@138.8.237.64:5060 Via: SIP/2.0/UDP 172.25.18.188:5060;branch=z9hG4bK14802cf2cc01 CSeq: 101 INVITE P-Preferred-Identity: Session-Expires: 1800;refresher=uac Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 2 IN IP4 172.25.18.188 s=SIP Call c=IN IP4 172.25.19.4 t=0 0 m=audio 25040 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (20 headers 10 lines) --- Sending to 172.25.18.188:5060 (no NAT) Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 172.25.19.4:25040 <--- Transmitting (no NAT) to 172.25.18.188:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.18.188:5060;branch=z9hG4bK14802cf2cc01;received=172.25.18.188 From: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753175 To: "Naidu, Abhishek" ;tag=as29868285 Call-ID: 724975d27232fc186ed7a7c80a34adbc@138.8.237.64:5060 CSeq: 101 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Length: 0 <------------> Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 172.25.18.188:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.25.18.188:5060;branch=z9hG4bK14802cf2cc01;received=172.25.18.188 From: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753175 To: "Naidu, Abhishek" ;tag=as29868285 Call-ID: 724975d27232fc186ed7a7c80a34adbc@138.8.237.64:5060 CSeq: 101 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: Content-Type: application/sdp Content-Length: 258 v=0 o=root 1403505908 1403505909 IN IP4 138.8.237.64 s=GS Asterisk PBX c=IN IP4 138.8.237.64 t=0 0 m=audio 30328 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:172.25.18.188:5060 ---> ACK sip:18366@138.8.237.64:5060 SIP/2.0 Date: Fri, 02 Dec 2011 07:34:15 GMT From: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753175 Allow-Events: presence, kpml Content-Length: 0 To: "Naidu, Abhishek" ;tag=as29868285 Call-ID: 724975d27232fc186ed7a7c80a34adbc@138.8.237.64:5060 Via: SIP/2.0/UDP 172.25.18.188:5060;branch=z9hG4bK148143b9ece8 CSeq: 101 ACK Max-Forwards: 70 <-------------> --- (10 headers 0 lines) --- == Manager 'ChatterBox' logged on from 10.221.196.232 == Spawn extension (macro-trunkdial, s, 2) exited non-zero on 'SIP/sip_trunk_primary-00000026' in macro 'trunkdial' == Spawn extension (callto-primary, 89991000, 2) exited non-zero on 'SIP/sip_trunk_primary-00000026' Scheduling destruction of SIP dialog '1e9d7cbc674045625fada08b3000cc1a@138.8.237.64:5060' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 172.25.18.188:5060 Reliably Transmitting (no NAT) to 172.25.18.188:5060: BYE sip:18366@172.25.18.188:5060 SIP/2.0 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK2515cb69 Max-Forwards: 70 From: "Naidu, Abhishek" ;tag=as271f7c90 To: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753173 Call-ID: 1e9d7cbc674045625fada08b3000cc1a@138.8.237.64:5060 CSeq: 103 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Dec 2 02:34:05] WARNING[28866]: channel.c:5769 __ast_channel_masquerade: Can't setup masquerade. One or both channels is dead. (Bridge/SIP/sip_trunk_primary-00000026 <-- SIP/sip_trunk_primary-00000026) == Manager 'ChatterBox' logged off from 10.221.196.232 <--- SIP read from UDP:172.25.18.188:5060 ---> SIP/2.0 200 OK Date: Fri, 02 Dec 2011 07:34:20 GMT From: "Naidu, Abhishek" ;tag=as271f7c90 Content-Length: 0 To: ;tag=4c3d5788-2975-4fff-a89b-90bcca02812a-30753173 Call-ID: 1e9d7cbc674045625fada08b3000cc1a@138.8.237.64:5060 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK2515cb69 CSeq: 103 BYE <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '1e9d7cbc674045625fada08b3000cc1a@138.8.237.64:5060' Method: INVITE nytadla01*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 172.25.18.188 89991000 724975d27232fc1 0x4 (ulaw) No Rx: ACK sip_trunk_ 1 active SIP dialog nytadla01*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 172.25.18.188 89991000 724975d27232fc1 0x4 (ulaw) No Rx: ACK sip_trunk_ 1 active SIP dialog Reliably Transmitting (no NAT) to 172.25.19.4:5060: OPTIONS sip:172.25.19.4 SIP/2.0 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK5df67ac5 Max-Forwards: 70 From: "asterisk" ;tag=as10be8310 To: Contact: Call-ID: 3e1d29c872d38cf07d8b60945bfc8c3f@138.8.237.64:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.7.1 Date: Fri, 02 Dec 2011 07:34:26 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:172.25.19.4:5060 ---> SIP/2.0 200 OK Date: Fri, 02 Dec 2011 07:34:26 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "asterisk" ;tag=as10be8310 Content-Length: 0 To: ;tag=1586083066 Call-ID: 3e1d29c872d38cf07d8b60945bfc8c3f@138.8.237.64:5060 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK5df67ac5 CSeq: 102 OPTIONS <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '3e1d29c872d38cf07d8b60945bfc8c3f@138.8.237.64:5060' Method: OPTIONS nytadla01*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 172.25.18.188 89991000 724975d27232fc1 0x4 (ulaw) No Rx: ACK sip_trunk_ 1 active SIP dialog nytadla01*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 172.25.18.188 89991000 724975d27232fc1 0x4 (ulaw) No Rx: ACK sip_trunk_ 1 active SIP dialog nytadla01*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 172.25.18.188 89991000 724975d27232fc1 0x4 (ulaw) No Rx: ACK sip_trunk_ 1 active SIP dialog Reliably Transmitting (no NAT) to 172.25.18.188:5060: OPTIONS sip:172.25.18.188 SIP/2.0 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK68f05731 Max-Forwards: 70 From: "asterisk" ;tag=as78723e2e To: Contact: Call-ID: 6b0661677c7839bd6d13208c04f6b0b9@138.8.237.64:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.7.1 Date: Fri, 02 Dec 2011 07:34:45 GMT Session-Expires: 1800 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:172.25.18.188:5060 ---> SIP/2.0 200 OK Date: Fri, 02 Dec 2011 07:35:00 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "asterisk" ;tag=as78723e2e Content-Length: 0 To: ;tag=1255665727 Call-ID: 6b0661677c7839bd6d13208c04f6b0b9@138.8.237.64:5060 Via: SIP/2.0/UDP 138.8.237.64:5060;branch=z9hG4bK68f05731 CSeq: 102 OPTIONS <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '6b0661677c7839bd6d13208c04f6b0b9@138.8.237.64:5060' Method: OPTIONS nytadla01*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 172.25.18.188 89991000 724975d27232fc1 0x4 (ulaw) No Rx: ACK sip_trunk_ 1 active SIP dialog nytadla01*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 172.25.18.188 89991000 724975d27232fc1 0x4 (ulaw) No Rx: ACK sip_trunk_ 1 active SIP dialog nytadla01*CLI> Disconnected from Asterisk server [astadmin@nytadla01 ~]$