rendered paste bodyasterisk1*CLI> sip debug
SIP Debugging enabled
asterisk1*CLI>
<-- SIP read from 192.168.2.167:5060:
INVITE sip:713092872137@192.168.2.6;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK4ed386b354825728
From: "Adam Shake" <sip:3101@192.168.2.6>;tag=519BB939-561A42AC
To: <sip:713092872137@192.168.2.6;user=phone>
CSeq: 1 INVITE
Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167
Contact: <sip:3101@192.168.2.167>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 251
v=0
o=- 1170843165 1170843165 IN IP4 192.168.2.167
s=Polycom IP Phone
c=IN IP4 192.168.2.167
t=0 0
a=sendrecv
m=audio 2240 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
--- (14 headers 11 lines) ---
Using INVITE request as basis request - aebd6655-3e46ebd7-f1173f7a@192.168.2.167
Sending to 192.168.2.167 : 5060 (non-NAT)
Reliably Transmitting (NAT) to 192.168.2.167:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK4ed386b354825728;received=192.168.2.167
From: "Adam Shake" <sip:3101@192.168.2.6>;tag=519BB939-561A42AC
To: <sip:713092872137@192.168.2.6;user=phone>;tag=as77dba809
Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="44e0293b"
Content-Length: 0
---
Scheduling destruction of call 'aebd6655-3e46ebd7-f1173f7a@192.168.2.167' in 15000 ms
Found user '3101'
asterisk1*CLI>
<-- SIP read from 192.168.2.167:5060:
ACK sip:713092872137@192.168.2.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK4ed386b354825728
From: "Adam Shake" <sip:3101@192.168.2.6>;tag=519BB939-561A42AC
To: <sip:713092872137@192.168.2.6;user=phone>;tag=as77dba809
CSeq: 1 ACK
Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167
Contact: <sip:3101@192.168.2.167>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041
Max-Forwards: 70
Content-Length: 0
--- (11 headers 0 lines) ---
asterisk1*CLI>
<-- SIP read from 192.168.2.167:5060:
INVITE sip:713092872137@192.168.2.6;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B
From: "Adam Shake" <sip:3101@192.168.2.6>;tag=519BB939-561A42AC
To: <sip:713092872137@192.168.2.6;user=phone>
CSeq: 2 INVITE
Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167
Contact: <sip:3101@192.168.2.167>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="3101", realm="asterisk", nonce="44e0293b", uri="sip:713092872137@192.168.2.6;user=phone", response="8a565ee944d6e131141968283cb9415c", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 251
v=0
o=- 1170843165 1170843165 IN IP4 192.168.2.167
s=Polycom IP Phone
c=IN IP4 192.168.2.167
t=0 0
a=sendrecv
m=audio 2240 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
--- (15 headers 11 lines) ---
Using INVITE request as basis request - aebd6655-3e46ebd7-f1173f7a@192.168.2.167
Sending to 192.168.2.167 : 5060 (NAT)
Found user '3101'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.167:2240
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 713092872137 in from-internal (domain 192.168.2.6;user=phone)
list_route: hop: <sip:3101@192.168.2.167>
Transmitting (NAT) to 192.168.2.167:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B;received=192.168.2.167
From: "Adam Shake" <sip:3101@192.168.2.6>;tag=519BB939-561A42AC
To: <sip:713092872137@192.168.2.6;user=phone>
Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:713092872137@192.168.2.6>
Content-Length: 0
---
-- Executing Macro("SIP/3101-09c10640", "dialout-trunk|1|13092872137||") in new stack
-- Executing Set("SIP/3101-09c10640", "DIAL_TRUNK=1") in new stack
-- Executing Set("SIP/3101-09c10640", "_NODEST=") in new stack
-- Executing Set("SIP/3101-09c10640", "DIAL_NUMBER=13092872137") in new stack
-- Executing Set("SIP/3101-09c10640", "ROUTE_PASSWD=") in new stack
-- Executing Set("SIP/3101-09c10640", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,8)
-- Executing Set("SIP/3101-09c10640", "GROUP()=OUT_1") in new stack
-- Executing Macro("SIP/3101-09c10640", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/3101-09c10640", "user-callerid: device 3101") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "0?report") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "0?start") in new stack
-- Executing Set("SIP/3101-09c10640", "REALCALLERIDNUM=3101") in new stack
-- Executing NoOp("SIP/3101-09c10640", "REALCALLERIDNUM is 3101") in new stack
-- Executing Set("SIP/3101-09c10640", "AMPUSER=3101") in new stack
-- Executing Set("SIP/3101-09c10640", "AMPUSERCIDNAME=Adam Shake") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "0?report") in new stack
-- Executing Set("SIP/3101-09c10640", "CALLERID(all)=Adam Shake <3101>") in new stack
-- Executing Set("SIP/3101-09c10640", "REALCALLERIDNUM=3101") in new stack
-- Executing NoOp("SIP/3101-09c10640", "TTL: ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/3101-09c10640", "Using CallerID "Adam Shake" <3101>") in new stack
-- Executing Macro("SIP/3101-09c10640", "record-enable|3101|OUT") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/3101-09c10640", "recordingcheck|20070207-214756|1170906476.2") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20070207-214756|1170906476.2: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/3101-09c10640", "No recording needed") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "0?skipoutcid") in new stack
-- Executing Set("SIP/3101-09c10640", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing Macro("SIP/3101-09c10640", "outbound-callerid|1") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/3101-09c10640", "REALCALLERIDNUM is 3101") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing Set("SIP/3101-09c10640", "USEROUTCID=") in new stack
-- Executing Set("SIP/3101-09c10640", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/3101-09c10640", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing GotoIf("SIP/3101-09c10640", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,18)
-- Executing GotoIf("SIP/3101-09c10640", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing NoOp("SIP/3101-09c10640", "CallerID set to "Adam Shake" <3101>") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,16)
-- Executing DeadAGI("SIP/3101-09c10640", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/3101-09c10640", "OUTNUM=13092872137") in new stack
-- Executing Set("SIP/3101-09c10640", "custom=ZAP/g0") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "0?customtrunk") in new stack
-- Executing Dial("SIP/3101-09c10640", "ZAP/g0/13092872137|300|") in new stack
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Goto("SIP/3101-09c10640", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing NoOp("SIP/3101-09c10640", "Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing Macro("SIP/3101-09c10640", "outisbusy|") in new stack
-- Executing Playback("SIP/3101-09c10640", "all-circuits-busy-now|noanswer") in new stack
We're at 192.168.2.6 port 16906
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 192.168.2.167:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B;received=192.168.2.167
From: "Adam Shake" <sip:3101@192.168.2.6>;tag=519BB939-561A42AC
To: <sip:713092872137@192.168.2.6;user=phone>;tag=as19fb306a
Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:713092872137@192.168.2.6>
Content-Type: application/sdp
Content-Length: 236
v=0
o=root 2536 2536 IN IP4 192.168.2.6
s=session
c=IN IP4 192.168.2.6
t=0 0
m=audio 16906 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 - - - -
---
-- Playing 'all-circuits-busy-now' (language 'en')
-- Executing Playback("SIP/3101-09c10640", "pls-try-call-later|noanswer") in new stack
-- Playing 'pls-try-call-later' (language 'en')
-- Executing Macro("SIP/3101-09c10640", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3101-09c10640", "w") in new stack
-- Executing NoCDR("SIP/3101-09c10640", "") in new stack
-- Executing GotoIf("SIP/3101-09c10640", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/3101-09c10640", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing Wait("SIP/3101-09c10640", "5") in new stack
asterisk1*CLI>
<-- SIP read from 192.168.2.167:5060:
CANCEL sip:713092872137@192.168.2.6;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B
From: "Adam Shake" <sip:3101@192.168.2.6>;tag=519BB939-561A42AC
To: <sip:713092872137@192.168.2.6;user=phone>
CSeq: 2 CANCEL
Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167
Contact: <sip:3101@192.168.2.167>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041
Proxy-Authorization: Digest username="3101", realm="asterisk", nonce="44e0293b", uri="sip:713092872137@192.168.2.6;user=phone", response="8a565ee944d6e131141968283cb9415c", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
--- (11 headers 0 lines) ---
Sending to 192.168.2.167 : 5060 (NAT)
Reliably Transmitting (NAT) to 192.168.2.167:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B;received=192.168.2.167
From: "Adam Shake" <sip:3101@192.168.2.6>;tag=519BB939-561A42AC
To: <sip:713092872137@192.168.2.6;user=phone>;tag=as19fb306a
Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
X-Asterisk-HangupCause: Circuit/channel congestion
---
Transmitting (NAT) to 192.168.2.167:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B;received=192.168.2.167
From: "Adam Shake" <sip:3101@192.168.2.6>;tag=519BB939-561A42AC
To: <sip:713092872137@192.168.2.6;user=phone>;tag=as19fb306a
Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:713092872137@192.168.2.6>
Content-Length: 0
X-Asterisk-HangupCause: Circuit/channel congestion
---
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3101-09c10640' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3101-09c10640' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3101-09c10640'
asterisk1*CLI>
<-- SIP read from 192.168.2.167:5060:
ACK sip:713092872137@192.168.2.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.167;branch=z9hG4bK42dfb2beF7CD507B
From: "Adam Shake" <sip:3101@192.168.2.6>;tag=519BB939-561A42AC
To: <sip:713092872137@192.168.2.6;user=phone>;tag=as19fb306a
CSeq: 2 ACK
Call-ID: aebd6655-3e46ebd7-f1173f7a@192.168.2.167
Contact: <sip:3101@192.168.2.167>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041
Max-Forwards: 70
Content-Length: 0
--- (11 headers 0 lines) ---
Destroying call 'aebd6655-3e46ebd7-f1173f7a@192.168.2.167'
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.2.167:5060:
OPTIONS sip:3101@192.168.2.167 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:5060;branch=z9hG4bK40aa3608;rport
From: "Unknown" <sip:Unknown@192.168.2.6>;tag=as7e8cbfdc
To: <sip:3101@192.168.2.167>
Contact: <sip:Unknown@192.168.2.6>
Call-ID: 07a98df94048464426ecac8770e27f2a@192.168.2.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 08 Feb 2007 03:48:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.2.167:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.6:5060;branch=z9hG4bK40aa3608;rport
From: "Unknown" <sip:Unknown@192.168.2.6>;tag=as7e8cbfdc
To: <sip:3101@192.168.2.167>;tag=57441B67-4D7B718A
CSeq: 102 OPTIONS
Call-ID: 07a98df94048464426ecac8770e27f2a@192.168.2.6
Contact: <sip:3101@192.168.2.167>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041
Content-Length: 0
--- (10 headers 0 lines) ---
Destroying call '07a98df94048464426ecac8770e27f2a@192.168.2.6'
asterisk1*CLI>