All pastes #76848 Raw Edit

darren

public text v1 · immutable
·published 2006-07-02 05:09 UTC
rendered paste body
To: <sip:0400518501@10.60.33.102;user=phone>
Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:309@10.60.33.9:2051;line=e68aihg7>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60i
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 364

v=0
o=root 1184052172 1184052172 IN IP4 10.60.33.9
s=call
c=IN IP4 10.60.33.9
t=0 0
m=audio 60380 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

--- (17 headers 17 lines)---
Using INVITE request as basis request - 3c41ee87a393-sdu3blgcu8bo@snom190
Sending to 10.60.33.9 : 2051 (NAT)
Found user '309'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 10.60.33.9:60380
Found description format pcmu
Found description format pcma
Found description format g722
Found description format g726-32
Found description format gsm
Found description format g729
Found description format g723
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|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 0400518501 in from-internal (domain 10.60.33.102)
list_route: hop: <sip:309@10.60.33.9:2051;line=e68aihg7>
Transmitting (no NAT) to 10.60.33.9:2051:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-v5320lemhqjc;rport;received=10.60.33.9
From: "Darren" <sip:309@10.60.33.102>;tag=bgybpgubi1
To: <sip:0400518501@10.60.33.102;user=phone>
Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0400518501@10.60.33.102>
Content-Length: 0


---
    -- Executing Macro("SIP/309-5000", "dialout-trunk|4|0400518501||") in new stack
    -- Executing GotoIf("SIP/309-5000", "1?3:2") in new stack
    -- Goto (macro-dialout-trunk,s,3)
    -- Executing Macro("SIP/309-5000", "user-callerid") in new stack
    -- Executing GotoIf("SIP/309-5000", "0?report") in new stack
    -- Executing GotoIf("SIP/309-5000", "0?start") in new stack
    -- Executing Set("SIP/309-5000", "REALCALLERIDNUM=309") in new stack
    -- Executing NoOp("SIP/309-5000", "REALCALLERIDNUM is 309") in new stack
    -- Executing Set("SIP/309-5000", "AMPUSER=309") in new stack
    -- Executing Set("SIP/309-5000", "AMPUSERCIDNAME=Darren") in new stack
    -- Executing GotoIf("SIP/309-5000", "0?report") in new stack
    -- Executing Set("SIP/309-5000", "CALLERID(all)=Darren <309>") in new stack
    -- Executing NoOp("SIP/309-5000", "Using CallerID "Darren" <309>") in new stack
    -- Executing Macro("SIP/309-5000", "record-enable|309|OUT") in new stack
    -- Executing GotoIf("SIP/309-5000", "0 > 0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing AGI("SIP/309-5000", "recordingcheck|20060702-130714|1151816834.76") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20060702-130714|1151816834.76: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/309-5000", "No recording needed") in new stack
    -- Executing Macro("SIP/309-5000", "outbound-callerid|4") in new stack
    -- Executing GotoIf("SIP/309-5000", "1?start") in new stack
    -- Goto (macro-outbound-callerid,s,3)
    -- Executing NoOp("SIP/309-5000", "REALCALLERIDNUM is 309") in new stack
    -- Executing Set("SIP/309-5000", "USEROUTCID=") in new stack
    -- Executing Set("SIP/309-5000", "EMERGENCYCID=") in new stack
    -- Executing Set("SIP/309-5000", "TRUNKOUTCID=") in new stack
    -- Executing GotoIf("SIP/309-5000", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,11)
    -- Executing GotoIf("SIP/309-5000", "1?usercid") in new stack
    -- Goto (macro-outbound-callerid,s,13)
    -- Executing GotoIf("SIP/309-5000", "1?report") in new stack
    -- Goto (macro-outbound-callerid,s,15)
    -- Executing NoOp("SIP/309-5000", "CallerID set to "Darren" <309>") in new stack
    -- Executing Set("SIP/309-5000", "GROUP()=OUT_4") in new stack
    -- Executing GotoIf("SIP/309-5000", "0?108") in new stack
    -- Executing Set("SIP/309-5000", "DIAL_NUMBER=0400518501") in new stack
    -- Executing Set("SIP/309-5000", "DIAL_TRUNK=4") in new stack
    -- Executing AGI("SIP/309-5000", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
  fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing Set("SIP/309-5000", "OUTNUM=0400518501") in new stack
    -- Executing Set("SIP/309-5000", "custom=ZAP/4") in new stack
    -- Executing GotoIf("SIP/309-5000", "0?16") in new stack
    -- Executing Dial("SIP/309-5000", "ZAP/4/0400518501|120|r") in new stack
    -- Called 4/0400518501
Transmitting (no NAT) to 10.60.33.9:2051:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-v5320lemhqjc;rport;received=10.60.33.9
From: "Darren" <sip:309@10.60.33.102>;tag=bgybpgubi1
To: <sip:0400518501@10.60.33.102;user=phone>;tag=as59bed056
Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0400518501@10.60.33.102>
Content-Length: 0


---
    -- Zap/4-1 answered SIP/309-5000
We're at 10.60.33.102 port 16944
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 10.60.33.9:2051:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-v5320lemhqjc;rport;received=10.60.33.9
From: "Darren" <sip:309@10.60.33.102>;tag=bgybpgubi1
To: <sip:0400518501@10.60.33.102;user=phone>;tag=as59bed056
Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0400518501@10.60.33.102>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 26441 26441 IN IP4 10.60.33.102
s=session
c=IN IP4 10.60.33.102
t=0 0
m=audio 16944 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 - - - -

---
trixboxdhn*CLI>
<-- SIP read from 10.60.33.9:2051:
ACK sip:0400518501@10.60.33.102 SIP/2.0
Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-uexurcnm06eu;rport
From: "Darren" <sip:309@10.60.33.102>;tag=bgybpgubi1
To: <sip:0400518501@10.60.33.102;user=phone>;tag=as59bed056
Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:309@10.60.33.9:2051;line=e68aihg7>
Content-Length: 0


--- (9 headers 0 lines)---
trixboxdhn*CLI>
<-- SIP read from 10.60.33.9:2051:
BYE sip:0400518501@10.60.33.102 SIP/2.0
Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-dsicr5hmakou;rport
From: "Darren" <sip:309@10.60.33.102>;tag=bgybpgubi1
To: <sip:0400518501@10.60.33.102;user=phone>;tag=as59bed056
Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190
CSeq: 2 BYE
Max-Forwards: 70
Contact: <sip:309@10.60.33.9:2051;line=e68aihg7>
User-Agent: snom190/3.60i
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 10.60.33.9 : 2051 (NAT)
Transmitting (NAT) to 10.60.33.9:2051:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-dsicr5hmakou;rport;received=10.60.33.9
From: "Darren" <sip:309@10.60.33.102>;tag=bgybpgubi1
To: <sip:0400518501@10.60.33.102;user=phone>;tag=as59bed056
Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0400518501@10.60.33.102>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing

trixboxdhn*CLI>
---
    -- Hungup 'Zap/4-1'
Destroying call '3c41ee87a393-sdu3blgcu8bo@snom190'
trixboxdhn*CLI>