All pastes #2058608 Raw Edit

Miscellany

public text v1 · immutable
#2058608 ·published 2011-05-13 23:07 UTC
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Reliably Transmitting (NAT) to 10.10.10.12:5060:
OPTIONS sip:203@10.10.10.12:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK6c10d46c;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as3188234b
To: <sip:203@10.10.10.12:5060>
Contact: <sip:Unknown@10.10.10.7>
Call-ID: 1ad1fc37175177e16bbb6edf1634986f@10.10.10.7
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.6.2.15)
Date: Fri, 13 May 2011 22:55:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.10.10.12:5060 --->
SIP/2.0 200 OK
To: <sip:203@10.10.10.12:5060>;tag=815d2a0b89199d2di0
From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as3188234b
Call-ID: 1ad1fc37175177e16bbb6edf1634986f@10.10.10.7
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK6c10d46c
Server: Linksys/SPA942-5.1.15(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1ad1fc37175177e16bbb6edf1634986f@10.10.10.7' Method: OPTIONS
Reliably Transmitting (NAT) to 10.10.10.13:5063:
OPTIONS sip:204@10.10.10.13:5063 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK60e43f14;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as47bd712b
To: <sip:204@10.10.10.13:5063>
Contact: <sip:Unknown@10.10.10.7>
Call-ID: 76b8ac3159bcc1f57cbb72c81527e709@10.10.10.7
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.6.2.15)
Date: Fri, 13 May 2011 22:55:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.10.10.13:5063 --->
SIP/2.0 200 OK
To: <sip:204@10.10.10.13:5063>;tag=47db739b438d60f6i3
From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as47bd712b
Call-ID: 76b8ac3159bcc1f57cbb72c81527e709@10.10.10.7
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK60e43f14
Server: Linksys/SPA942-5.1.15(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '76b8ac3159bcc1f57cbb72c81527e709@10.10.10.7' Method: OPTIONS
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 216.58.0.51:5060:
REGISTER sip:216.58.0.51 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK10e4284a;rport
Max-Forwards: 70
From: <sip:6477232824@216.58.0.51>;tag=as731aa619
To: <sip:6477232824@216.58.0.51>
Call-ID: 7f310bc557a68d50120b4df406a1cb25@127.0.0.1
CSeq: 156 REGISTER
User-Agent: FPBX-2.8.1(1.6.2.15)
Expires: 120
Contact: <sip:6477232824@10.10.10.7>
Content-Length: 0


---
Really destroying SIP dialog '7f310bc557a68d50120b4df406a1cb25@127.0.0.1' Method: REGISTER
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/203-0000067c", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/203-0000067c", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/203-0000067c", "RC=111") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/203-0000067c", "111,1") in new stack
    -- Goto (macro-dialout-trunk,111,1)
    -- Executing [111@macro-dialout-trunk:1] Goto("SIP/203-0000067c", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/203-0000067c", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/203-0000067c", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 111 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/203-0000067c", "CALLERID(number)=203") in new stack
    -- Executing [14168853558@from-internal:7] Macro("SIP/203-0000067c", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/203-0000067c", "") in new stack
Audio is at 10.10.10.7 port 13208
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 10.10.10.12:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.10.10.12:5060;branch=z9hG4bK-6cac2822;received=10.10.10.12
From: "Itamar Desk" <sip:203@10.10.10.7>;tag=c170e01fdb5309eeo0
To: <sip:14168853558@10.10.10.7>;tag=as5164f61c
Call-ID: 5f0b6cb3-40cdc66a@10.10.10.12
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.6.2.15)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:14168853558@10.10.10.7>
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 831236311 831236311 IN IP4 10.10.10.7
s=Asterisk PBX 1.6.2.15
c=IN IP4 10.10.10.7
t=0 0
m=audio 13208 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=ptime:20
a=sendrecv

<------------>
    -- Executing [s@macro-outisbusy:2] Playback("SIP/203-0000067c", "all-circuits-busy-now,noanswer") in new stack
    -- <SIP/203-0000067c> Playing 'all-circuits-busy-now.gsm' (language 'en')
Reliably Transmitting (no NAT) to 216.58.0.51:5060:
OPTIONS sip:216.58.0.51 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK3e6a1076;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as55da66c0
To: <sip:216.58.0.51>
Contact: <sip:Unknown@10.10.10.7>
Call-ID: 25d039fe0b3275196af086f236208a45@10.10.10.7
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.6.2.15)
Date: Fri, 13 May 2011 22:55:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '7d58cc153c55f77a17992a5c1a0bb70d@cia.com' Method: INVITE

<--- SIP read from UDP:216.58.0.51:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK3e6a1076;received=173.35.236.158;rport=5060
From: "Unknown" <sip:Unknown@10.10.10.7>;tag=as55da66c0
To: <sip:216.58.0.51>;tag=as5c249a62
Call-ID: 25d039fe0b3275196af086f236208a45@10.10.10.7
CSeq: 102 OPTIONS
User-Agent: CIA.com PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '25d039fe0b3275196af086f236208a45@10.10.10.7' Method: OPTIONS
Retransmitting #1 (no NAT) to 216.58.0.51:5060:
REGISTER sip:216.58.0.51 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK10e4284a;rport
Max-Forwards: 70
From: <sip:6477232824@216.58.0.51>;tag=as731aa619
To: <sip:6477232824@216.58.0.51>
Call-ID: 7f310bc557a68d50120b4df406a1cb25@127.0.0.1
CSeq: 156 REGISTER
User-Agent: FPBX-2.8.1(1.6.2.15)
Expires: 120
Contact: <sip:6477232824@10.10.10.7>
Content-Length: 0


---
Retransmitting #2 (no NAT) to 216.58.0.51:5060:
REGISTER sip:216.58.0.51 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.7:5060;branch=z9hG4bK10e4284a;rport
Max-Forwards: 70
From: <sip:6477232824@216.58.0.51>;tag=as731aa619
To: <sip:6477232824@216.58.0.51>
Call-ID: 7f310bc557a68d50120b4df406a1cb25@127.0.0.1
CSeq: 156 REGISTER
User-Agent: FPBX-2.8.1(1.6.2.15)
Expires: 120
Contact: <sip:6477232824@10.10.10.7>
Content-Length: 0


---
    -- Executing [s@macro-outisbusy:3] Playback("SIP/203-0000067c", "pls-try-call-later,noanswer") in new stack
    -- <SIP/203-0000067c> Playing 'pls-try-call-later.gsm' (language 'en')
pbx*CLI>