All pastes #180293 Raw Edit

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#180293 ·published 2006-09-23 00:19 UTC
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Using INVITE request as basis request - 577c4eb4-ee8554f6-8a319bff@192.168.0.7
Sending to 192.168.0.7: 5060; transport UDP (non-NAT)
Found peer '102'
Reliably Transmitting (no NAT) to 192.168.0.7:5060:UDP
 with fd 15
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK869709106BE141CB;received=192.168.0.7
From: "Front Desk" <sip:102@192.168.0.6>;tag=839BB768-7EBCC249
To: <sip:101@192.168.0.6;user=phone>;tag=as17d35f9e
Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7
CSeq: 3 INVITE
User-Agent: ATA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:101@192.168.0.6;transport=UDP>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40f6f98c"
Content-Length: 0


---
UDP write: fd 15
Scheduling destruction of call '577c4eb4-ee8554f6-8a319bff@192.168.0.7' in 15000 ms
asterisk1*CLI>
<-- Sip read from 192.168.0.7:5060:UDP
ACK sip:101@192.168.0.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK869709106BE141CB
From: "Front Desk" <sip:102@192.168.0.6>;tag=839BB768-7EBCC249
To: <sip:101@192.168.0.6;user=phone>;tag=as17d35f9e
CSeq: 3 ACK
Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7
Contact: <sip:102@192.168.0.7>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.3.0019
Max-Forwards: 70
Content-Length: 0


--- (11 headers 0 lines)---
asterisk1*CLI>
<-- Sip read from 192.168.0.7:5060:UDP
INVITE sip:101@192.168.0.6;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK3cec9bf270E15735
From: "Front Desk" <sip:102@192.168.0.6>;tag=839BB768-7EBCC249
To: <sip:101@192.168.0.6;user=phone>
CSeq: 4 INVITE
Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7
Contact: <sip:102@192.168.0.7>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.3.0019
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="102", realm="asterisk", nonce="40f6f98c", uri="sip:101@192.168.0.6;user=phone", response="8adc9ca5c6942ae325328deb151c13d9", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 247

v=0
o=- 1158970534 1158970534 IN IP4 192.168.0.7
s=Polycom IP Phone
c=IN IP4 192.168.0.7
t=0 0
a=sendrecv
m=audio 2228 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

--- (15 headers 11 lines)---
Using INVITE request as basis request - 577c4eb4-ee8554f6-8a319bff@192.168.0.7
Sending to 192.168.0.7: 5060; transport UDP (non-NAT)
Found peer '102'
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 192.168.0.7:2228
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 101 in from-internal (domain 192.168.0.6;user=phone)
list_route: hop: <sip:102@192.168.0.7>
Transmitting (no NAT) to 192.168.0.7:5060:UDP
 with fd 15
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK3cec9bf270E15735;received=192.168.0.7
From: "Front Desk" <sip:102@192.168.0.6>;tag=839BB768-7EBCC249
To: <sip:101@192.168.0.6;user=phone>
Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7
CSeq: 4 INVITE
User-Agent: ATA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:101@192.168.0.6;transport=UDP>
Content-Length: 0


---
UDP write: fd 15
    -- Executing Macro("SIP/102-08848eb0", "exten-vm|101|101") in new stack
    -- Executing Macro("SIP/102-08848eb0", "user-callerid") in new stack
    -- Executing GotoIf("SIP/102-08848eb0", "0?report") in new stack
    -- Executing GotoIf("SIP/102-08848eb0", "0?start") in new stack
    -- Executing Set("SIP/102-08848eb0", "REALCALLERIDNUM=102") in new stack
    -- Executing NoOp("SIP/102-08848eb0", "REALCALLERIDNUM is 102") in new stack
    -- Executing Set("SIP/102-08848eb0", "AMPUSER=102") in new stack
    -- Executing Set("SIP/102-08848eb0", "AMPUSERCIDNAME=Front Desk") in new stack
    -- Executing GotoIf("SIP/102-08848eb0", "0?report") in new stack
    -- Executing Set("SIP/102-08848eb0", "CALLERID(all)=Front Desk <102>") in new stack
    -- Executing NoOp("SIP/102-08848eb0", "Using CallerID "Front Desk" <102>") in new stack
    -- Executing Set("SIP/102-08848eb0", "FROMCONTEXT=exten-vm") in new stack
    -- Executing Set("SIP/102-08848eb0", "VMBOX=101") in new stack
    -- Executing Set("SIP/102-08848eb0", "EXTTOCALL=101") in new stack
    -- Executing Set("SIP/102-08848eb0", "CFUEXT=") in new stack
    -- Executing Set("SIP/102-08848eb0", "RT=15") in new stack
    -- Executing Macro("SIP/102-08848eb0", "record-enable|101|IN") in new stack
    -- Executing GotoIf("SIP/102-08848eb0", "0 > 0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing AGI("SIP/102-08848eb0", "recordingcheck|20060922-191540|1158970540.5") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20060922-191540|1158970540.5: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/102-08848eb0", "No recording needed") in new stack
    -- Executing GotoIf("SIP/102-08848eb0", "0?dolocaldial|1") in new stack
    -- Executing Macro("SIP/102-08848eb0", "dial|15|tr|101") in new stack
    -- Executing AGI("SIP/102-08848eb0", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Starting New Dialparties.agi
    --  dialparties.agi: priority is 1
  dialparties.agi: Caller ID name is 'Front Desk' number is '102'
  dialparties.agi: Methodology of ring is  'none'
    --  dialparties.agi: Added extension 101 to extension map
    --  dialparties.agi: Extension 101 cf is disabled
    --  dialparties.agi: Extension 101 do not disturb is disabled
       >  dialparties.agi: extnum: 101
       >  dialparties.agi: exthascw: 0
       >  dialparties.agi: exthascfb: 0
       >  dialparties.agi: extcfb:
       >  dialparties.agi: exthascfu: 0
       >  dialparties.agi: extcfu:
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  == Manager 'admin' logged off from 127.0.0.1
       >  dialparties.agi: ExtensionState: 0
    --  dialparties.agi: Checking CW and CFB status for extension 101
    --  dialparties.agi: DbSet CALLTRACE/101 to 102
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing Dial("SIP/102-08848eb0", "SIP/101|15|tr") in new stack
UDP write: fd 15
    -- Called 101
Transmitting (no NAT) to 192.168.0.7:5060:UDP
 with fd 15
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK3cec9bf270E15735;received=192.168.0.7
From: "Front Desk" <sip:102@192.168.0.6>;tag=839BB768-7EBCC249
To: <sip:101@192.168.0.6;user=phone>;tag=as76448c74
Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7
CSeq: 4 INVITE
User-Agent: ATA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:101@192.168.0.6;transport=UDP>
Content-Length: 0


---
UDP write: fd 15
    -- SIP/101-088546e8 is ringing
UDP write: fd 15
    -- SIP/101-088546e8 answered SIP/102-08848eb0
UDP write: fd 15
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-08848eb0' in macro 'dial'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-08848eb0' in macro 'exten-vm'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-08848eb0'
Reliably Transmitting (no NAT) to 192.168.0.7:5060:UDP
 with fd 15
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK3cec9bf270E15735;received=192.168.0.7
From: "Front Desk" <sip:102@192.168.0.6>;tag=839BB768-7EBCC249
To: <sip:101@192.168.0.6;user=phone>;tag=as76448c74
Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7
CSeq: 4 INVITE
User-Agent: ATA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:101@192.168.0.6;transport=UDP>
Content-Length: 0


---
UDP write: fd 15
asterisk1*CLI>
<-- Sip read from 192.168.0.7:5060:UDP
ACK sip:101@192.168.0.6;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.7;branch=z9hG4bK3cec9bf270E15735
From: "Front Desk" <sip:102@192.168.0.6>;tag=839BB768-7EBCC249
To: <sip:101@192.168.0.6;user=phone>;tag=as76448c74
CSeq: 4 ACK
Call-ID: 577c4eb4-ee8554f6-8a319bff@192.168.0.7
Contact: <sip:102@192.168.0.7>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.3.0019
Max-Forwards: 70
Content-Length: 0


--- (11 headers 0 lines)---
Destroying call '577c4eb4-ee8554f6-8a319bff@192.168.0.7'