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- Really destroying SIP dialog 'NjI4Mjg4OWRjMDVhYzlkNDEyYTc0ZmUzMWY5YTg4ODM.' Method: REGISTER
- <--- SIP read from UDP:192.168.6.14:64428 --->
- INVITE sip:19163777000@192.168.6.16:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-7452910378389766-1---d8754z-;rport
- Max-Forwards: 70
- Contact: <sip:4002@10.0.0.2:64428;rinstance=a0c537b71d7010f7>
- To: <sip:19163777000@192.168.6.16:5060>
- From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
- Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
- CSeq: 1 INVITE
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
- Content-Type: application/sdp
- Supported: replaces
- User-Agent: 3CXPhone 6.0.26523.0
- Content-Length: 391
- v=0
- o=3cxVCE 344141565 37306590 IN IP4 10.0.0.2
- s=3cxVCE Audio Call
- c=IN IP4 10.0.0.2
- t=0 0
- m=audio 40012 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=ptime:20
- a=sendrecv
- m=video 40010 RTP/AVP 34
- c=IN IP4 10.0.0.2
- a=rtpmap:34 H263/90000
- a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
- a=sendrecv
- <------------->
- --- (13 headers 18 lines) ---
- Sending to 192.168.6.14:64428 (NAT)
- Using INVITE request as basis request - MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
- Found peer '4002' for '4002' from 192.168.6.14:64428
- <--- Reliably Transmitting (NAT) to 192.168.6.14:64428 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-7452910378389766-1---d8754z-;received=192.168.6.14;rport=64428
- From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
- To: <sip:19163777000@192.168.6.16:5060>;tag=as4a5d64ee
- Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
- CSeq: 1 INVITE
- Server: FPBX-2.8.1(1.8.11.0)
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4110095b"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog 'MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.' in 6848 ms (Method: INVITE)
- <--- SIP read from UDP:192.168.6.14:64428 --->
- ACK sip:19163777000@192.168.6.16:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-7452910378389766-1---d8754z-;rport
- Max-Forwards: 70
- To: <sip:19163777000@192.168.6.16:5060>;tag=as4a5d64ee
- From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
- Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
- CSeq: 1 ACK
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- <--- SIP read from UDP:192.168.6.14:64428 --->
- INVITE sip:19163777000@192.168.6.16:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-9f4ae525885ede12-1---d8754z-;rport
- Max-Forwards: 70
- Contact: <sip:4002@10.0.0.2:64428;rinstance=a0c537b71d7010f7>
- To: <sip:19163777000@192.168.6.16:5060>
- From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
- Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
- CSeq: 2 INVITE
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
- Content-Type: application/sdp
- Supported: replaces
- User-Agent: 3CXPhone 6.0.26523.0
- Authorization: Digest username="4002",realm="asterisk",nonce="4110095b",uri="sip:19163777000@192.168.6.16:5060",response="c0ebdeed27eeef0a84c8df4ea47d5824",algorithm=MD5
- Content-Length: 391
- v=0
- o=3cxVCE 344141565 37306590 IN IP4 10.0.0.2
- s=3cxVCE Audio Call
- c=IN IP4 10.0.0.2
- t=0 0
- m=audio 40012 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=ptime:20
- a=sendrecv
- m=video 40010 RTP/AVP 34
- c=IN IP4 10.0.0.2
- a=rtpmap:34 H263/90000
- a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
- a=sendrecv
- <------------->
- --- (14 headers 18 lines) ---
- Sending to 192.168.6.14:64428 (NAT)
- Using INVITE request as basis request - MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
- Found peer '4002' for '4002' from 192.168.6.14:64428
- == Using SIP RTP TOS bits 184
- == Using SIP RTP CoS mark 5
- Found RTP audio format 0
- Found RTP audio format 8
- Found RTP audio format 3
- Found RTP audio format 101
- Found audio description format PCMU for ID 0
- Found audio description format PCMA for ID 8
- Found audio description format GSM for ID 3
- Found audio description format telephone-event for ID 101
- Found RTP video format 34
- Found video description format H263 for ID 34
- Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/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 10.0.0.2:40012
- Looking for 19163777000 in from-internal (domain 192.168.6.16)
- list_route: hop: <sip:4002@10.0.0.2:64428;rinstance=a0c537b71d7010f7>
- <--- Transmitting (NAT) to 192.168.6.14:64428 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-9f4ae525885ede12-1---d8754z-;received=192.168.6.14;rport=64428
- From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
- To: <sip:19163777000@192.168.6.16:5060>
- Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
- CSeq: 2 INVITE
- Server: FPBX-2.8.1(1.8.11.0)
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:19163777000@192.168.6.16:5060>
- Content-Length: 0
- <------------>
- -- Executing [19163777000@from-internal:1] Macro("SIP/4002-00000054", "user-callerid,SKIPTTL,") in new stack
- -- Executing [s@macro-user-callerid:1] Set("SIP/4002-00000054", "AMPUSER=4002") in new stack
- -- Executing [s@macro-user-callerid:2] GotoIf("SIP/4002-00000054", "0?report") in new stack
- -- Executing [s@macro-user-callerid:3] ExecIf("SIP/4002-00000054", "1?Set(REALCALLERIDNUM=4002)") in new stack
- -- Executing [s@macro-user-callerid:4] Set("SIP/4002-00000054", "AMPUSER=4002") in new stack
- -- Executing [s@macro-user-callerid:5] Set("SIP/4002-00000054", "AMPUSERCIDNAME=4002") in new stack
- -- Executing [s@macro-user-callerid:6] GotoIf("SIP/4002-00000054", "0?report") in new stack
- -- Executing [s@macro-user-callerid:7] Set("SIP/4002-00000054", "AMPUSERCID=4002") in new stack
- -- Executing [s@macro-user-callerid:8] Set("SIP/4002-00000054", "CALLERID(all)="4002" <4002>") in new stack
- -- Executing [s@macro-user-callerid:9] ExecIf("SIP/4002-00000054", "0?Set(CHANNEL(language)=)") in new stack
- -- Executing [s@macro-user-callerid:10] GotoIf("SIP/4002-00000054", "1?continue") in new stack
- -- Goto (macro-user-callerid,s,19)
- -- Executing [s@macro-user-callerid:19] Set("SIP/4002-00000054", "CALLERID(number)=4002") in new stack
- -- Executing [s@macro-user-callerid:20] Set("SIP/4002-00000054", "CALLERID(name)=4002") in new stack
- -- Executing [s@macro-user-callerid:21] NoOp("SIP/4002-00000054", "Using CallerID "4002" <4002>") in new stack
- -- Executing [19163777000@from-internal:2] NoOp("SIP/4002-00000054", "Calling Out Route: nymgo") in new stack
- -- Executing [19163777000@from-internal:3] Set("SIP/4002-00000054", "MOHCLASS=default") in new stack
- -- Executing [19163777000@from-internal:4] Set("SIP/4002-00000054", "_NODEST=") in new stack
- -- Executing [19163777000@from-internal:5] Macro("SIP/4002-00000054", "record-enable,4002,OUT,") in new stack
- -- Executing [s@macro-record-enable:1] GotoIf("SIP/4002-00000054", "1?check") in new stack
- -- Goto (macro-record-enable,s,4)
- -- Executing [s@macro-record-enable:4] ExecIf("SIP/4002-00000054", "0?MacroExit()") in new stack
- -- Executing [s@macro-record-enable:5] GotoIf("SIP/4002-00000054", "0?Group:OUT") in new stack
- -- Goto (macro-record-enable,s,15)
- -- Executing [s@macro-record-enable:15] GotoIf("SIP/4002-00000054", "0?IN") in new stack
- -- Executing [s@macro-record-enable:16] ExecIf("SIP/4002-00000054", "1?MacroExit()") in new stack
- -- Executing [19163777000@from-internal:6] Macro("SIP/4002-00000054", "dialout-trunk,3,19163777000,") in new stack
- -- Executing [s@macro-dialout-trunk:1] Set("SIP/4002-00000054", "DIAL_TRUNK=3") in new stack
- -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/4002-00000054", "0?sub-pincheck,s,1") in new stack
- -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/4002-00000054", "0?disabletrunk,1") in new stack
- -- Executing [s@macro-dialout-trunk:4] Set("SIP/4002-00000054", "DIAL_NUMBER=19163777000") in new stack
- -- Executing [s@macro-dialout-trunk:5] Set("SIP/4002-00000054", "DIAL_TRUNK_OPTIONS=tr") in new stack
- -- Executing [s@macro-dialout-trunk:6] Set("SIP/4002-00000054", "OUTBOUND_GROUP=OUT_3") in new stack
- -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/4002-00000054", "1?nomax") in new stack
- -- Goto (macro-dialout-trunk,s,9)
- -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/4002-00000054", "0?skipoutcid") in new stack
- -- Executing [s@macro-dialout-trunk:10] Set("SIP/4002-00000054", "DIAL_TRUNK_OPTIONS=") in new stack
- -- Executing [s@macro-dialout-trunk:11] Macro("SIP/4002-00000054", "outbound-callerid,3") in new stack
- -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/4002-00000054", "0?Set(CALLERPRES()=)") in new stack
- -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/4002-00000054", "0?Set(REALCALLERIDNUM=4002)") in new stack
- -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/4002-00000054", "1?normcid") in new stack
- -- Goto (macro-outbound-callerid,s,6)
- -- Executing [s@macro-outbound-callerid:6] Set("SIP/4002-00000054", "USEROUTCID=") in new stack
- -- Executing [s@macro-outbound-callerid:7] Set("SIP/4002-00000054", "EMERGENCYCID=") in new stack
- -- Executing [s@macro-outbound-callerid:8] Set("SIP/4002-00000054", "TRUNKOUTCID=94771771229") in new stack
- -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/4002-00000054", "1?trunkcid") in new stack
- -- Goto (macro-outbound-callerid,s,12)
- -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/4002-00000054", "1?Set(CALLERID(all)=94771771229)") in new stack
- -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/4002-00000054", "0?Set(CALLERID(all)=)") in new stack
- -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/4002-00000054", "0?Set(CALLERID(all)=)") in new stack
- -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/4002-00000054", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
- -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/4002-00000054", "1?sub-flp-3,s,1") in new stack
- -- Executing [s@sub-flp-3:1] ExecIf("SIP/4002-00000054", "1?Return()") in new stack
- -- Executing [s@macro-dialout-trunk:13] Set("SIP/4002-00000054", "OUTNUM=19163777000") in new stack
- -- Executing [s@macro-dialout-trunk:14] Set("SIP/4002-00000054", "custom=SIP/com-in") in new stack
- -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/4002-00000054", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
- -- Executing [s@macro-dialout-trunk:16] Macro("SIP/4002-00000054", "dialout-trunk-predial-hook,") in new stack
- -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/4002-00000054", "") in new stack
- -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/4002-00000054", "0?bypass,1") in new stack
- -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/4002-00000054", "0?customtrunk") in new stack
- -- Executing [s@macro-dialout-trunk:19] Dial("SIP/4002-00000054", "SIP/com-in/19163777000,300,") in new stack
- == Using SIP RTP TOS bits 184
- == Using SIP RTP CoS mark 5
- Audio is at 17482
- Adding codec 0x4 (ulaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 176.34.131.112:5060:
- INVITE sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK57b4a304;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 102 INVITE
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Date: Mon, 10 Sep 2012 19:40:53 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 235
- v=0
- o=root 620339841 620339841 IN IP4 192.168.6.16
- s=Asterisk PBX 1.8.11.0
- c=IN IP4 192.168.6.16
- t=0 0
- m=audio 17482 RTP/AVP 0 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- -- Called SIP/com-in/19163777000
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 100 Trying...
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK57b4a304;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 102 INVITE
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK57b4a304;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.8853
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 102 INVITE
- Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e426300000083ee05bbbdd6b252cdf7c847987f9adb05"
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- set_destination: Parsing <sip:19163777000@sip.commpeak.com> for address/port to send to
- set_destination: set destination to 176.34.131.112:5060
- Transmitting (NAT) to 176.34.131.112:5060:
- ACK sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK57b4a304;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.8853
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 102 ACK
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Content-Length: 0
- ---
- Audio is at 17482
- Adding codec 0x4 (ulaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 176.34.131.112:5060:
- INVITE sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK00ed031b;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 103 INVITE
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e426300000083ee05bbbdd6b252cdf7c847987f9adb05", response="4164d3b6a47684856459808e7f47093a"
- Date: Mon, 10 Sep 2012 19:40:53 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 235
- v=0
- o=root 620339841 620339842 IN IP4 192.168.6.16
- s=Asterisk PBX 1.8.11.0
- c=IN IP4 192.168.6.16
- t=0 0
- m=audio 17482 RTP/AVP 0 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- Retransmitting #1 (NAT) to 176.34.131.112:5060:
- INVITE sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK00ed031b;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 103 INVITE
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e426300000083ee05bbbdd6b252cdf7c847987f9adb05", response="4164d3b6a47684856459808e7f47093a"
- Date: Mon, 10 Sep 2012 19:40:53 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 235
- v=0
- o=root 620339841 620339842 IN IP4 192.168.6.16
- s=Asterisk PBX 1.8.11.0
- c=IN IP4 192.168.6.16
- t=0 0
- m=audio 17482 RTP/AVP 0 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 100 Trying...
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK00ed031b;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 103 INVITE
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK00ed031b;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.48be
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 103 INVITE
- Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e4263000000842f1b6d8bd6345ce0b4141bcb0e2d7d46"
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- set_destination: Parsing <sip:19163777000@sip.commpeak.com> for address/port to send to
- set_destination: set destination to 176.34.131.112:5060
- Transmitting (NAT) to 176.34.131.112:5060:
- ACK sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK00ed031b;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.48be
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 103 ACK
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Content-Length: 0
- ---
- Audio is at 17482
- Adding codec 0x4 (ulaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 176.34.131.112:5060:
- INVITE sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK6c464bd0;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 104 INVITE
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e4263000000842f1b6d8bd6345ce0b4141bcb0e2d7d46", response="cfb11196a40bdc8a55e5be2f8a0ad95f"
- Date: Mon, 10 Sep 2012 19:40:53 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 235
- v=0
- o=root 620339841 620339843 IN IP4 192.168.6.16
- s=Asterisk PBX 1.8.11.0
- c=IN IP4 192.168.6.16
- t=0 0
- m=audio 17482 RTP/AVP 0 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 100 Trying...
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK00ed031b;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 103 INVITE
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK00ed031b;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.48be
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 103 INVITE
- Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e426400000085bafbe6e7191759dc8717a00903df722e"
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- Transmitting (NAT) to 176.34.131.112:5060:
- ACK sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK6c464bd0;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 103 ACK
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Content-Length: 0
- ---
- Retransmitting #1 (NAT) to 176.34.131.112:5060:
- INVITE sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK6c464bd0;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 104 INVITE
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e4263000000842f1b6d8bd6345ce0b4141bcb0e2d7d46", response="cfb11196a40bdc8a55e5be2f8a0ad95f"
- Date: Mon, 10 Sep 2012 19:40:53 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 235
- v=0
- o=root 620339841 620339843 IN IP4 192.168.6.16
- s=Asterisk PBX 1.8.11.0
- c=IN IP4 192.168.6.16
- t=0 0
- m=audio 17482 RTP/AVP 0 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 100 Trying...
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK6c464bd0;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 104 INVITE
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK6c464bd0;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.f3c7
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 104 INVITE
- Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e4264000000867504ae8f83abb84f8a735fffd2382c7a"
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- Transmitting (NAT) to 176.34.131.112:5060:
- ACK sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK6c464bd0;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.f3c7
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 104 ACK
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Content-Length: 0
- ---
- Audio is at 17482
- Adding codec 0x4 (ulaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 176.34.131.112:5060:
- INVITE sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK5648f608;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 105 INVITE
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e4264000000867504ae8f83abb84f8a735fffd2382c7a", response="6931a37ce53b7d79e79e84e61f5c74d3"
- Date: Mon, 10 Sep 2012 19:40:54 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 235
- v=0
- o=root 620339841 620339844 IN IP4 192.168.6.16
- s=Asterisk PBX 1.8.11.0
- c=IN IP4 192.168.6.16
- t=0 0
- m=audio 17482 RTP/AVP 0 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 100 Trying...
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK6c464bd0;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 104 INVITE
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK6c464bd0;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.f3c7
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 104 INVITE
- Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e426400000087eda82a7a2bb10253eb730e3b40451d7a"
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- Transmitting (NAT) to 176.34.131.112:5060:
- ACK sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK5648f608;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 104 ACK
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Content-Length: 0
- ---
- Retransmitting #1 (NAT) to 176.34.131.112:5060:
- INVITE sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK5648f608;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 105 INVITE
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e4264000000867504ae8f83abb84f8a735fffd2382c7a", response="6931a37ce53b7d79e79e84e61f5c74d3"
- Date: Mon, 10 Sep 2012 19:40:54 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 235
- v=0
- o=root 620339841 620339844 IN IP4 192.168.6.16
- s=Asterisk PBX 1.8.11.0
- c=IN IP4 192.168.6.16
- t=0 0
- m=audio 17482 RTP/AVP 0 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 100 Trying...
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK5648f608;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 105 INVITE
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK5648f608;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.6012
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 105 INVITE
- Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e42640000008868309f249a1d20138aaae9bd40454093"
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- Transmitting (NAT) to 176.34.131.112:5060:
- ACK sip:19163777000@sip.commpeak.com SIP/2.0
- Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK5648f608;rport
- Max-Forwards: 70
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.6012
- Contact: <sip:94771771229@192.168.6.16:5060>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 105 ACK
- User-Agent: FPBX-2.8.1(1.8.11.0)
- Content-Length: 0
- ---
- -- SIP/com-in-00000055 is circuit-busy
- == Everyone is busy/congested at this time (1:0/1/0)
- -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/4002-00000054", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
- -- Executing [s@macro-dialout-trunk:21] Goto("SIP/4002-00000054", "s-CONGESTION,1") in new stack
- -- Goto (macro-dialout-trunk,s-CONGESTION,1)
- -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/4002-00000054", "RC=21") in new stack
- -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/4002-00000054", "21,1") in new stack
- -- Goto (macro-dialout-trunk,21,1)
- -- Executing [21@macro-dialout-trunk:1] Goto("SIP/4002-00000054", "continue,1") in new stack
- -- Goto (macro-dialout-trunk,continue,1)
- -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/4002-00000054", "1?noreport") in new stack
- -- Goto (macro-dialout-trunk,continue,3)
- -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/4002-00000054", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack
- -- Executing [continue@macro-dialout-trunk:4] Set("SIP/4002-00000054", "CALLERID(number)=4002") in new stack
- -- Executing [19163777000@from-internal:7] Macro("SIP/4002-00000054", "outisbusy,") in new stack
- -- Executing [s@macro-outisbusy:1] Progress("SIP/4002-00000054", "") in new stack
- Audio is at 18978
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x8 (alaw) to SDP
- Adding codec 0x2 (gsm) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- <--- Transmitting (NAT) to 192.168.6.14:64428 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-9f4ae525885ede12-1---d8754z-;received=192.168.6.14;rport=64428
- From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
- To: <sip:19163777000@192.168.6.16:5060>;tag=as53a48ffe
- Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
- CSeq: 2 INVITE
- Server: FPBX-2.8.1(1.8.11.0)
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:19163777000@192.168.6.16:5060>
- Content-Type: application/sdp
- Content-Length: 302
- v=0
- o=root 66937535 66937535 IN IP4 192.168.6.16
- s=Asterisk PBX 1.8.11.0
- c=IN IP4 192.168.6.16
- t=0 0
- m=audio 18978 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- m=video 0 RTP/AVP 34
- <------------>
- -- Executing [s@macro-outisbusy:2] GotoIf("SIP/4002-00000054", "0?emergency,1") in new stack
- -- Executing [s@macro-outisbusy:3] GotoIf("SIP/4002-00000054", "0?intracompany,1") in new stack
- -- Executing [s@macro-outisbusy:4] Playback("SIP/4002-00000054", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
- -- <SIP/4002-00000054> Playing 'all-circuits-busy-now.gsm' (language 'en')
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 100 Trying...
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK5648f608;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 105 INVITE
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- Really destroying SIP dialog '783561206945f8c3671514bc125c60e1@192.168.6.16:5060' Method: INVITE
- <--- SIP read from UDP:176.34.131.112:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK5648f608;rport=4831
- From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
- To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.6012
- Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
- CSeq: 105 INVITE
- Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e42640000008991566720346b100b76969387973b0cb9"
- Server: CommPeak SIP Proxy
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- -- <SIP/4002-00000054> Playing 'pls-try-call-later.gsm' (language 'en')
- <--- SIP read from UDP:192.168.6.14:64428 --->
- <------------->
- -- Executing [s@macro-outisbusy:5] Congestion("SIP/4002-00000054", "20") in new stack
- <--- Reliably Transmitting (NAT) to 192.168.6.14:64428 --->
- SIP/2.0 503 Service Unavailable
- Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-9f4ae525885ede12-1---d8754z-;received=192.168.6.14;rport=64428
- From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
- To: <sip:19163777000@192.168.6.16:5060>;tag=as53a48ffe
- Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
- CSeq: 2 INVITE
- Server: FPBX-2.8.1(1.8.11.0)
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- X-Asterisk-HangupCause: Call Rejected
- X-Asterisk-HangupCauseCode: 21
- Content-Length: 0
- <------------>
- == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/4002-00000054' in macro 'outisbusy'
- == Spawn extension (from-internal, 19163777000, 7) exited non-zero on 'SIP/4002-00000054'
- -- Executing [h@from-internal:1] Macro("SIP/4002-00000054", "hangupcall") in new stack
- -- Executing [s@macro-hangupcall:1] GotoIf("SIP/4002-00000054", "1?endmixmoncheck") in new stack
- -- Goto (macro-hangupcall,s,9)
- -- Executing [s@macro-hangupcall:9] NoOp("SIP/4002-00000054", "End of MIXMON check") in new stack
- -- Executing [s@macro-hangupcall:10] GotoIf("SIP/4002-00000054", "1?nomeetmemon") in new stack
- -- Goto (macro-hangupcall,s,28)
- -- Executing [s@macro-hangupcall:28] NoOp("SIP/4002-00000054", "End of MEETME check") in new stack
- -- Executing [s@macro-hangupcall:29] GotoIf("SIP/4002-00000054", "1?noautomon") in new stack
- -- Goto (macro-hangupcall,s,34)
- -- Executing [s@macro-hangupcall:34] NoOp("SIP/4002-00000054", "TOUCH_MONITOR_OUTPUT=") in new stack
- -- Executing [s@macro-hangupcall:35] GotoIf("SIP/4002-00000054", "1?noautomon2") in new stack
- -- Goto (macro-hangupcall,s,41)
- -- Executing [s@macro-hangupcall:41] NoOp("SIP/4002-00000054", "MONITOR_FILENAME=") in new stack
- -- Executing [s@macro-hangupcall:42] GotoIf("SIP/4002-00000054", "1?skiprg") in new stack
- -- Goto (macro-hangupcall,s,45)
- -- Executing [s@macro-hangupcall:45] GotoIf("SIP/4002-00000054", "1?skipblkvm") in new stack
- -- Goto (macro-hangupcall,s,48)
- -- Executing [s@macro-hangupcall:48] GotoIf("SIP/4002-00000054", "1?theend") in new stack
- -- Goto (macro-hangupcall,s,50)
- -- Executing [s@macro-hangupcall:50] AGI("SIP/4002-00000054", "hangup.agi") in new stack
- -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
- <--- SIP read from UDP:192.168.6.14:64428 --->
- ACK sip:19163777000@192.168.6.16:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-9f4ae525885ede12-1---d8754z-;rport
- Max-Forwards: 70
- To: <sip:19163777000@192.168.6.16:5060>;tag=as53a48ffe
- From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
- Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
- CSeq: 2 ACK
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- -- <SIP/4002-00000054>AGI Script hangup.agi completed, returning 0
- -- Executing [s@macro-hangupcall:51] Hangup("SIP/4002-00000054", "") in new stack
- == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/4002-00000054' in macro 'hangupcall'
- == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/4002-00000054'
- Really destroying SIP dialog 'MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.' Method: ACK
- <--- SIP read from UDP:61.245.172.12:7035 --->
- <------------->
- pbx*CLI> sip set debug off
- SIP Debugging Disabled
- pbx*CLI>
- pbx*CLI> pbx*CLI>
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