rendered paste bodyasterisk1*CLI> sip debug
SIP Debugging re-enabled
Destroying call '50f21849787b1a5623e030a903be9e57@127.0.0.1'
asterisk1*CLI>
<-- SIP read from 192.168.11.111:63318:
--- (0 headers 1 lines) ---
Destroying call '5f0400ab79c91d7a1ae64fe763e60f9d@192.168.11.112'
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.11.113:5060:
OPTIONS sip:64@192.168.11.113:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK339ae831;rport
From: "Unknown" <sip:Unknown@192.168.11.112>;tag=as36206ca3
To: <sip:64@192.168.11.113:5060>
Contact: <sip:Unknown@192.168.11.112>
Call-ID: 2ce61a537ed698641c2b59d44dc4b934@192.168.11.112
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 09 Jul 2007 13:23:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.11.113:5060:
SIP/2.0 200 OK
To: <sip:64@192.168.11.113:5060>;tag=d8edacfaa725db8ci0
From: "Unknown" <sip:Unknown@192.168.11.112>;tag=as36206ca3
Call-ID: 2ce61a537ed698641c2b59d44dc4b934@192.168.11.112
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK339ae831
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
--- (10 headers 0 lines) ---
Destroying call '2ce61a537ed698641c2b59d44dc4b934@192.168.11.112'
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.11.113:5060:
OPTIONS sip:62@192.168.11.113:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK46009d6b;rport
From: "Unknown" <sip:Unknown@192.168.11.112>;tag=as4900cf24
To: <sip:62@192.168.11.113:5060>
Contact: <sip:Unknown@192.168.11.112>
Call-ID: 7656b1bc1ce5ceb90c8f26201e8a41e1@192.168.11.112
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 09 Jul 2007 13:23:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.11.113:5060:
SIP/2.0 200 OK
To: <sip:62@192.168.11.113:5060>;tag=f45df18a6757e03ci1
From: "Unknown" <sip:Unknown@192.168.11.112>;tag=as4900cf24
Call-ID: 7656b1bc1ce5ceb90c8f26201e8a41e1@192.168.11.112
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK46009d6b
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
--- (10 headers 0 lines) ---
Destroying call '7656b1bc1ce5ceb90c8f26201e8a41e1@192.168.11.112'
asterisk1*CLI>
<-- SIP read from 192.168.11.114:5060:
INVITE sip:8*675148048384@192.168.11.112;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.11.114:5060
Max-Forwards: 70
From: 63 <sip:63@192.168.11.112;user=phone>;tag=e6204a9553045f22
To: <sip:8*675148048384@192.168.11.112;user=phone>
Call-ID: 945cbc7a903ed9d4@192.168.11.114
CSeq: 1 INVITE
Contact: 63 <sip:63@192.168.11.114:5060;user=phone>
Supported: timer
Session-Expires: 180
Content-Type: application/sdp
Content-Length: 194
v=0
o=- 1 0 IN IP4 192.168.11.114
s=-
c=IN IP4 192.168.11.114
t=0 0
m=audio 49152 RTP/AVP 18 0 8 2
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726/8000
--- (12 headers 10 lines) ---
Using INVITE request as basis request - 945cbc7a903ed9d4@192.168.11.114
Sending to 192.168.11.114 : 5060 (non-NAT)
Reliably Transmitting (NAT) to 192.168.11.114:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114
From: 63 <sip:63@192.168.11.112;user=phone>;tag=e6204a9553045f22
To: <sip:8*675148048384@192.168.11.112;user=phone>;tag=as67579b5b
Call-ID: 945cbc7a903ed9d4@192.168.11.114
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ece17b0"
Content-Length: 0
---
Scheduling destruction of call '945cbc7a903ed9d4@192.168.11.114' in 15000 ms
Found user '63'
asterisk1*CLI>
<-- SIP read from 192.168.11.114:5060:
ACK sip:8*675148048384@192.168.11.112;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.11.114:5060
Max-Forwards: 70
From: 63 <sip:63@192.168.11.112;user=phone>;tag=e6204a9553045f22
To: <sip:8*675148048384@192.168.11.112;user=phone>;tag=as67579b5b
Call-ID: 945cbc7a903ed9d4@192.168.11.114
CSeq: 1 ACK
Content-Length: 0
--- (8 headers 0 lines) ---
asterisk1*CLI>
<-- SIP read from 192.168.11.114:5060:
INVITE sip:8*675148048384@192.168.11.112;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.11.114:5060
Max-Forwards: 70
From: 63 <sip:63@192.168.11.112;user=phone>;tag=e6204a9553045f22
To: <sip:8*675148048384@192.168.11.112;user=phone>
Call-ID: 945cbc7a903ed9d4@192.168.11.114
CSeq: 2 INVITE
Contact: 63 <sip:63@192.168.11.114:5060;user=phone>
Supported: timer
Session-Expires: 180
Proxy-Authorization: Digest username="63", realm="asterisk", nonce="1ece17b0", opaque="", uri="sip:8*675148048384@192.168.11.112", response="cca8a821db56cf20691e0322216f4782"
Content-Type: application/sdp
Content-Length: 194
v=0
o=- 1 0 IN IP4 192.168.11.114
s=-
c=IN IP4 192.168.11.114
t=0 0
m=audio 49152 RTP/AVP 18 0 8 2
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726/8000
--- (13 headers 10 lines) ---
Using INVITE request as basis request - 945cbc7a903ed9d4@192.168.11.114
Sending to 192.168.11.114 : 5060 (NAT)
Found user '63'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Peer audio RTP is at port 192.168.11.114:49152
Found description format G729
Found description format PCMU
Found description format PCMA
Found description format G726
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 8*675148048384 in from-internal (domain 192.168.11.112;user=phone)
list_route: hop: <sip:63@192.168.11.114:5060;user=phone>
Transmitting (NAT) to 192.168.11.114:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114
From: 63 <sip:63@192.168.11.112;user=phone>;tag=e6204a9553045f22
To: <sip:8*675148048384@192.168.11.112;user=phone>
Call-ID: 945cbc7a903ed9d4@192.168.11.114
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8*675148048384@192.168.11.112>
Content-Length: 0
---
-- Executing Macro("SIP/63-09ab1cc8", "dialout-trunk|3|*675148048384||") in new stack
-- Executing Set("SIP/63-09ab1cc8", "DIAL_TRUNK=3") in new stack
-- Executing Set("SIP/63-09ab1cc8", "_NODEST=") in new stack
-- Executing Set("SIP/63-09ab1cc8", "DIAL_NUMBER=*675148048384") in new stack
-- Executing Set("SIP/63-09ab1cc8", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/63-09ab1cc8", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,8)
-- Executing Set("SIP/63-09ab1cc8", "GROUP()=OUT_3") in new stack
-- Executing Macro("SIP/63-09ab1cc8", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/63-09ab1cc8", "user-callerid: device 63") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "0?report") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "0?start") in new stack
-- Executing Set("SIP/63-09ab1cc8", "REALCALLERIDNUM=63") in new stack
-- Executing NoOp("SIP/63-09ab1cc8", "REALCALLERIDNUM is 63") in new stack
-- Executing Set("SIP/63-09ab1cc8", "AMPUSER=63") in new stack
-- Executing Set("SIP/63-09ab1cc8", "AMPUSERCIDNAME=Catherine Laplante") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "0?report") in new stack
-- Executing Set("SIP/63-09ab1cc8", "CALLERID(all)=Catherine Laplante <63>") in new stack
-- Executing Set("SIP/63-09ab1cc8", "REALCALLERIDNUM=63") in new stack
-- Executing NoOp("SIP/63-09ab1cc8", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/63-09ab1cc8", "Using CallerID "Catherine Laplante" <63>") in new stack
-- Executing Macro("SIP/63-09ab1cc8", "record-enable|63|OUT") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/63-09ab1cc8", "recordingcheck|20070709-092340|1183987420.164") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20070709-092340|1183987420.164: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/63-09ab1cc8", "No recording needed") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "0?skipoutcid") in new stack
-- Executing Set("SIP/63-09ab1cc8", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/63-09ab1cc8", "outbound-callerid|3") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/63-09ab1cc8", "REALCALLERIDNUM is 63") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/63-09ab1cc8", "USEROUTCID=") in new stack
-- Executing Set("SIP/63-09ab1cc8", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/63-09ab1cc8", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/63-09ab1cc8", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/63-09ab1cc8", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/63-09ab1cc8", "CallerID set to "Catherine Laplante" <63>") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "0?nomax") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "0?chanfull") in new stack
-- Executing DeadAGI("SIP/63-09ab1cc8", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/63-09ab1cc8", "OUTNUM=*675148048384") in new stack
-- Executing Set("SIP/63-09ab1cc8", "custom=SIP/Acanac2") in new stack
-- Executing GotoIf("SIP/63-09ab1cc8", "0?customtrunk") in new stack
-- Executing Dial("SIP/63-09ab1cc8", "SIP/Acanac2/*675148048384|300|") in new stack
We're at 192.168.11.112 port 11418
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 66.49.255.38:5060:
INVITE sip:*675148048384@66.49.255.38 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK09d899b2;rport
From: "Catherine Laplante" <sip:63@192.168.11.112>;tag=as4bb6a3eb
To: <sip:*675148048384@66.49.255.38>
Contact: <sip:63@192.168.11.112>
Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 09 Jul 2007 13:23:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 5118 5118 IN IP4 192.168.11.112
s=session
c=IN IP4 192.168.11.112
t=0 0
m=audio 11418 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 - - - -
---
-- Called Acanac2/*675148048384
asterisk1*CLI>
<-- SIP read from 66.49.255.38:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 24.200.182.199:5060;branch=z9hG4bK09d899b2;received=24.200.182.199;rport=5060
From: "Catherine Laplante" <sip:63@192.168.11.112>;tag=as4bb6a3eb
To: <sip:*675148048384@66.49.255.38>;tag=as49c94d71
Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="62b3a69a"
Content-Length: 0
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 66.49.255.38:5060:
ACK sip:*675148048384@66.49.255.38 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK09d899b2;rport
From: "Catherine Laplante" <sip:63@192.168.11.112>;tag=as4bb6a3eb
To: <sip:*675148048384@66.49.255.38>;tag=as49c94d71
Contact: <sip:63@192.168.11.112>
Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
We're at 192.168.11.112 port 11418
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 66.49.255.38:5060:
INVITE sip:*675148048384@66.49.255.38 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK58cfe938;rport
From: "Catherine Laplante" <sip:63@192.168.11.112>;tag=as4bb6a3eb
To: <sip:*675148048384@66.49.255.38>
Contact: <sip:63@192.168.11.112>
Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="5146674639", realm="asterisk", algorithm=MD5, uri="sip:*675148048384@66.49.255.38", nonce="62b3a69a", response="863d0ebfd04278bc0a05f05575f37eb9", opaque=""
Date: Mon, 09 Jul 2007 13:23:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 5118 5119 IN IP4 192.168.11.112
s=session
c=IN IP4 192.168.11.112
t=0 0
m=audio 11418 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 - - - -
---
asterisk1*CLI>
<-- SIP read from 66.49.255.38:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 24.200.182.199:5060;branch=z9hG4bK58cfe938;received=24.200.182.199;rport=5060
From: "Catherine Laplante" <sip:63@192.168.11.112>;tag=as4bb6a3eb
To: <sip:*675148048384@66.49.255.38>;tag=as49c94d71
Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 66.49.255.38:5060:
ACK sip:*675148048384@66.49.255.38 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK58cfe938;rport
From: "Catherine Laplante" <sip:63@192.168.11.112>;tag=as4bb6a3eb
To: <sip:*675148048384@66.49.255.38>;tag=as49c94d71
Contact: <sip:63@192.168.11.112>
Call-ID: 3bb1953f4c35d621612dd8db4031d986@192.168.11.112
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/Acanac2-09b2b6d8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Goto("SIP/63-09ab1cc8", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing NoOp("SIP/63-09ab1cc8", "Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing Macro("SIP/63-09ab1cc8", "outisbusy|") in new stack
-- Executing Playback("SIP/63-09ab1cc8", "all-circuits-busy-now|noanswer") in new stack
We're at 192.168.11.112 port 12858
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Transmitting (NAT) to 192.168.11.114:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114
From: 63 <sip:63@192.168.11.112;user=phone>;tag=e6204a9553045f22
To: <sip:8*675148048384@192.168.11.112;user=phone>;tag=as496c2560
Call-ID: 945cbc7a903ed9d4@192.168.11.114
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8*675148048384@192.168.11.112>
Content-Type: application/sdp
Content-Length: 186
v=0
o=root 5118 5118 IN IP4 192.168.11.112
s=session
c=IN IP4 192.168.11.112
t=0 0
m=audio 12858 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
---
-- Playing 'all-circuits-busy-now' (language 'fr')
Destroying call '3bb1953f4c35d621612dd8db4031d986@192.168.11.112'
asterisk1*CLI>
<-- SIP read from 66.49.255.42:5060:
OPTIONS sip:5146674473@24.200.182.199:5060 SIP/2.0
Via: SIP/2.0/UDP 66.49.255.42:5060;branch=z9hG4bK5d59fe42;rport
From: "asterisk" <sip:asterisk@66.49.255.42>;tag=as02b5530a
To: <sip:5146674473@24.200.182.199:5060>
Contact: <sip:asterisk@66.49.255.42>
Call-ID: 53a75b0f743ed9e7421a29ef6255a81b@66.49.255.42
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 08 Jul 2007 17:52:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
--- (13 headers 0 lines) ---
Looking for 5146674473 in from-trunk (domain 24.200.182.199)
Transmitting (no NAT) to 66.49.255.42:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.49.255.42:5060;branch=z9hG4bK5d59fe42;received=66.49.255.42;rport=5060
From: "asterisk" <sip:asterisk@66.49.255.42>;tag=as02b5530a
To: <sip:5146674473@24.200.182.199:5060>;tag=as787d3d86
Call-ID: 53a75b0f743ed9e7421a29ef6255a81b@66.49.255.42
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.11.112>
Accept: application/sdp
Content-Length: 0
---
Destroying call '53a75b0f743ed9e7421a29ef6255a81b@66.49.255.42'
-- Executing Playback("SIP/63-09ab1cc8", "pls-try-call-later|noanswer") in new stack
-- Playing 'pls-try-call-later' (language 'fr')
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.11.114:5060:
OPTIONS sip:63@192.168.11.114:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK5253eae1;rport
From: "Unknown" <sip:Unknown@192.168.11.112>;tag=as2e7eef05
To: <sip:63@192.168.11.114:5060;user=phone>
Contact: <sip:Unknown@192.168.11.112>
Call-ID: 3233bd1c492363d178281c0e12b16a16@192.168.11.112
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 09 Jul 2007 13:23:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.11.114:5060:
CANCEL sip:8*675148048384@192.168.11.112;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.11.114:5060
Max-Forwards: 70
From: 63 <sip:63@192.168.11.112;user=phone>;tag=e6204a9553045f22
To: <sip:8*675148048384@192.168.11.112;user=phone>
Call-ID: 945cbc7a903ed9d4@192.168.11.114
CSeq: 2 CANCEL
Content-Length: 0
--- (8 headers 0 lines) ---
Sending to 192.168.11.114 : 5060 (NAT)
Reliably Transmitting (NAT) to 192.168.11.114:5060:
SIP/2.0 487 Request Terminated
ia: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114
From: 63 <sip:63@192.168.11.112;user=phone>;tag=e6204a9553045f22
To: <sip:8*675148048384@192.168.11.112;user=phone>;tag=as496c2560
Call-ID: 945cbc7a903ed9d4@192.168.11.114
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
X-Asterisk-HangupCause: Unallocated (unassigned) number
---
Transmitting (NAT) to 192.168.11.114:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.114:5060;received=192.168.11.114
From: 63 <sip:63@192.168.11.112;user=phone>;tag=e6204a9553045f22
To: <sip:8*675148048384@192.168.11.112;user=phone>;tag=as496c2560
Call-ID: 945cbc7a903ed9d4@192.168.11.114
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8*675148048384@192.168.11.112>
Content-Length: 0
X-Asterisk-HangupCause: Unallocated (unassigned) number
---
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/63-09ab1cc8' in macro 'outisbusy'
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/63-09ab1cc8'
asterisk1*CLI>
<-- SIP read from 192.168.11.114:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.112:5060;branch=z9hG4bK5253eae1;rport=
From: "Unknown" <sip:Unknown@192.168.11.112>;tag=as2e7eef05
To: <sip:63@192.168.11.114:5060;user=phone>;tag=95f741a83b81a040
Call-ID: 3233bd1c492363d178281c0e12b16a16@192.168.11.112
CSeq: 102 OPTIONS
Content-Type: application/sdp
Content-Length: 190
v=0
o=- 1 0 IN IP4 192.168.11.114
s=-
c=IN IP4 192.168.11.114
t=0 0
m=audio 0 RTP/AVP 18 0 8 2
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726/8000
--- (8 headers 10 lines) ---
Destroying call '3233bd1c492363d178281c0e12b16a16@192.168.11.112'
asterisk1*CLI>
<-- SIP read from 192.168.11.114:5060:
ACK sip:8*675148048384@192.168.11.112;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.11.114:5060
Max-Forwards: 70
From: 63 <sip:63@192.168.11.112;user=phone>;tag=e6204a9553045f22
To: <sip:8*675148048384@192.168.11.112;user=phone>;tag=as496c2560
Call-ID: 945cbc7a903ed9d4@192.168.11.114
CSeq: 2 ACK
Content-Length: 0
--- (8 headers 0 lines) ---
Destroying call '945cbc7a903ed9d4@192.168.11.114'
asterisk1*CLI> quit