Public feed #892872

Mine

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·published 2008-02-05 19:12 UTC
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Connected to Asterisk 1.4.17-1 RPM by vc-rpms@voipconsulting.nl currently running on trixbox1 (pid = 2570)
Verbosity is at least 12
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_additional.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
    -- Executing [090222902@from-internal:1] Macro("SIP/1000-091f5490", "dialout-trunk|1|90222902||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/1000-091f5490", "DIAL_TRUNK=1") in new stack
    -- Executing [s@macro-dialout-trunk:2] Set("SIP/1000-091f5490", "DIAL_NUMBER=90222902") in new stack
    -- Executing [s@macro-dialout-trunk:3] Set("SIP/1000-091f5490", "ROUTE_PASSWD=") in new stack
    -- Executing [s@macro-dialout-trunk:4] GotoIf("SIP/1000-091f5490", "1?noauth") in new stack
    -- Goto (macro-dialout-trunk,s,6)
    -- Executing [s@macro-dialout-trunk:6] GotoIf("SIP/1000-091f5490", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:7] Set("SIP/1000-091f5490", "_NODEST=") in new stack
    -- Executing [s@macro-dialout-trunk:8] Set("SIP/1000-091f5490", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:9] Set("SIP/1000-091f5490", "GROUP()=OUT_1") in new stack
    -- Executing [s@macro-dialout-trunk:10] Macro("SIP/1000-091f5490", "user-callerid|SKIPTTL") in new stack
    -- Executing [s@macro-user-callerid:1] NoOp("SIP/1000-091f5490", "user-callerid: device 1000") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/1000-091f5490", "AMPUSER=1000") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/1000-091f5490", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] GotoIf("SIP/1000-091f5490", "0?start") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/1000-091f5490", "REALCALLERIDNUM=1000") in new stack
    -- Executing [s@macro-user-callerid:6] NoOp("SIP/1000-091f5490", "REALCALLERIDNUM is 1000") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/1000-091f5490", "AMPUSER=1000") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/1000-091f5490", "AMPUSERCIDNAME=Nicolai Solem HOME") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/1000-091f5490", "0?report") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/1000-091f5490", "AMPUSERCID=1000") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/1000-091f5490", "CALLERID(all)="Nicolai Solem HOME" <1000>") in new stack
    -- Executing [s@macro-user-callerid:12] Set("SIP/1000-091f5490", "REALCALLERIDNUM=1000") in new stack
    -- Executing [s@macro-user-callerid:13] NoOp("SIP/1000-091f5490", "TTL:  ARG1: SKIPTTL") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("SIP/1000-091f5490", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,23)
    -- Executing [s@macro-user-callerid:23] NoOp("SIP/1000-091f5490", "Using CallerID "Nicolai Solem HOME" <1000>") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1000-091f5490", "record-enable|1000|OUT") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/1000-091f5490", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/1000-091f5490", "recordingcheck|20080205-200906|1202238546.456") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20080205-200906|1202238546.456: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] NoOp("SIP/1000-091f5490", "No recording needed") in new stack
    -- Executing [s@macro-dialout-trunk:12] GotoIf("SIP/1000-091f5490", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/1000-091f5490", "DIAL_TRUNK_OPTIONS=TW") in new stack
    -- Executing [s@macro-dialout-trunk:14] Macro("SIP/1000-091f5490", "outbound-callerid|1") in new stack
    -- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/1000-091f5490", "1?start") in new stack
    -- Goto (macro-outbound-callerid,s,3)
    -- Executing [s@macro-outbound-callerid:3] NoOp("SIP/1000-091f5490", "REALCALLERIDNUM is 1000") in new stack
    -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/1000-091f5490", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,9)
    -- Executing [s@macro-outbound-callerid:9] Set("SIP/1000-091f5490", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:10] Set("SIP/1000-091f5490", "EMERGENCYCID=<73614>") in new stack
    -- Executing [s@macro-outbound-callerid:11] Set("SIP/1000-091f5490", "TRUNKOUTCID=<73614>") in new stack
    -- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/1000-091f5490", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,16)
    -- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/1000-091f5490", "0?usercid") in new stack
    -- Executing [s@macro-outbound-callerid:17] Set("SIP/1000-091f5490", "CALLERID(all)=<73614>") in new stack
    -- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/1000-091f5490", "1?report") in new stack
    -- Goto (macro-outbound-callerid,s,22)
    -- Executing [s@macro-outbound-callerid:22] NoOp("SIP/1000-091f5490", "CallerID set to "" <73614>") in new stack
    -- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/1000-091f5490", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,17)
    -- Executing [s@macro-dialout-trunk:17] AGI("SIP/1000-091f5490", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing [s@macro-dialout-trunk:18] Set("SIP/1000-091f5490", "OUTNUM=90222902") in new stack
    -- Executing [s@macro-dialout-trunk:19] Set("SIP/1000-091f5490", "custom=SIP/CBK") in new stack
    -- Executing [s@macro-dialout-trunk:20] GotoIf("SIP/1000-091f5490", "1?gocall") in new stack
    -- Goto (macro-dialout-trunk,s,22)
    -- Executing [s@macro-dialout-trunk:22] Macro("SIP/1000-091f5490", "dialout-trunk-predial-hook") in new stack
    -- Executing [s@macro-dialout-trunk:23] GotoIf("SIP/1000-091f5490", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/1000-091f5490", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:25] Dial("SIP/1000-091f5490", "SIP/CBK/90222902|300|TW") in new stack
Audio is at 91.135.36.85 port 14664
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Reliably Transmitting (NAT) to 88.87.41.169:5060:
INVITE sip:90222902@sip.cbktele.com SIP/2.0
Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK220c9756;rport
From: "73614" <sip:73614@91.135.36.85>;tag=as35d4b006
To: <sip:90222902@sip.cbktele.com>
Contact: <sip:73614@91.135.36.85>
Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 05 Feb 2008 19:09:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 2570 2570 IN IP4 91.135.36.85
s=session
c=IN IP4 91.135.36.85
t=0 0
m=audio 14664 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called CBK/90222902
trixbox1*CLI>
<--- SIP read from 88.87.41.169:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK220c9756;received=91.135.36.85;rport=5060
From: "73614" <sip:73614@91.135.36.85>;tag=as35d4b006
To: <sip:90222902@sip.cbktele.com>;tag=as342a6cbd
Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85
CSeq: 102 INVITE
User-Agent: CBK Telecom as Sip 0.9.50
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="260ff3e8"
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 88.87.41.169:5060:
ACK sip:90222902@sip.cbktele.com SIP/2.0
Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK220c9756;rport
From: "73614" <sip:73614@91.135.36.85>;tag=as35d4b006
To: <sip:90222902@sip.cbktele.com>;tag=as342a6cbd
Contact: <sip:73614@91.135.36.85>
Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 91.135.36.85 port 14664
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Reliably Transmitting (NAT) to 88.87.41.169:5060:
INVITE sip:90222902@sip.cbktele.com SIP/2.0
Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK728d2005;rport
From: "73614" <sip:73614@91.135.36.85>;tag=as35d4b006
To: <sip:90222902@sip.cbktele.com>
Contact: <sip:73614@91.135.36.85>
Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="73614", realm="asterisk", algorithm=MD5, uri="sip:90222902@sip.cbktele.com", nonce="260ff3e8", response="ca723957914e486213037a3b0a50ec5d", opaque=""
Date: Tue, 05 Feb 2008 19:09:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 2570 2571 IN IP4 91.135.36.85
s=session
c=IN IP4 91.135.36.85
t=0 0
m=audio 14664 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
trixbox1*CLI>
<--- SIP read from 88.87.41.169:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK728d2005;received=91.135.36.85;rport=5060
From: "73614" <sip:73614@91.135.36.85>;tag=as35d4b006
To: <sip:90222902@sip.cbktele.com>
Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85
CSeq: 103 INVITE
User-Agent: CBK Telecom as Sip 0.9.50
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:90222902@88.87.41.169>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---

<--- SIP read from 88.87.41.169:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK728d2005;received=91.135.36.85;rport=5060
From: "73614" <sip:73614@91.135.36.85>;tag=as35d4b006
To: <sip:90222902@sip.cbktele.com>;tag=as19604792
Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85
CSeq: 103 INVITE
User-Agent: CBK Telecom as Sip 0.9.50
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:90222902@88.87.41.169>
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 22909 22909 IN IP4 88.87.41.169
s=session
c=IN IP4 88.87.41.169
t=0 0
m=audio 16392 RTP/AVP 8 3 0
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

<------------->
--- (11 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Peer audio RTP is at port 88.87.41.169:16392
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 88.87.41.169:16392
list_route: hop: <sip:90222902@88.87.41.169>
set_destination: Parsing <sip:90222902@88.87.41.169> for address/port to send to
set_destination: set destination to 88.87.41.169, port 5060
Transmitting (NAT) to 88.87.41.169:5060:
ACK sip:90222902@88.87.41.169 SIP/2.0
Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK55028cab;rport
From: "73614" <sip:73614@91.135.36.85>;tag=as35d4b006
To: <sip:90222902@sip.cbktele.com>;tag=as19604792
Contact: <sip:73614@91.135.36.85>
Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/CBK-09176620 answered SIP/1000-091f5490
Scheduling destruction of SIP dialog '1f914f3076a49fad32815ea9394f33dd@91.135.36.85' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:90222902@88.87.41.169> for address/port to send to
set_destination: set destination to 88.87.41.169, port 5060
Reliably Transmitting (NAT) to 88.87.41.169:5060:
BYE sip:90222902@88.87.41.169 SIP/2.0
Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK632720cf;rport
From: "73614" <sip:73614@91.135.36.85>;tag=as35d4b006
To: <sip:90222902@sip.cbktele.com>;tag=as19604792
Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="73614", realm="asterisk", algorithm=MD5, uri="sip:90222902@88.87.41.169", nonce="260ff3e8", response="10fdbd78ec963af3ec24c29858014275", opaque=""
Content-Length: 0


---
  == Spawn extension (macro-dialout-trunk, s, 25) exited non-zero on 'SIP/1000-091f5490' in macro 'dialout-trunk'
  == Spawn extension (macro-dialout-trunk, s, 25) exited non-zero on 'SIP/1000-091f5490'
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/1000-091f5490", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1000-091f5490", "w") in new stack
trixbox1*CLI>
<--- SIP read from 88.87.41.169:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK632720cf;received=91.135.36.85;rport=5060
From: "73614" <sip:73614@91.135.36.85>;tag=as35d4b006
To: <sip:90222902@sip.cbktele.com>;tag=as19604792
Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85
CSeq: 104 BYE
User-Agent: CBK Telecom as Sip 0.9.50
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:90222902@88.87.41.169>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1f914f3076a49fad32815ea9394f33dd@91.135.36.85' Method: INVITE
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/1000-091f5490", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/1000-091f5490", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/1000-091f5490", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/1000-091f5490", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/1000-091f5490", "") in new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1000-091f5490' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1000-091f5490'
Really destroying SIP dialog '644fe4dd2fee1498213c35055713507c@127.0.0.1' Method: REGISTER