Hi All,
I was using asterisk 1.6(ubuntu 10.04) and I can not remember whether i have added mysql addson or not ,
recently i have upgraded my os from ubuntu 10.04 to 11.04 and by default asterisk has been upgraded to 1.8.3
I followed the instruction of ‘adnanraza’(his first reply) to the post forums.asterisk.org/viewtopic.php?f=1&t=69484
but no cdr has been saved to the table.
How can i check whether my installed asterisk supports cdr recording into mysql db ?
Is saving cdr to mysql table is done automatically as long as you have cdr adds on or should be done manually by by adding some lines of code inside dailplan/agiscript ?
sip set debug on
SIP Debugging enabled
== Manager 'manager' logged on from 127.0.0.1
== Using SIP RTP CoS mark 5
We think we can do text
Audio is at 5060
Adding codec 0x40 (slin) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.2:5061:
INVITE sip:ivan@192.168.0.2:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK548003d0
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as5430b682
To: <sip:ivan@192.168.0.2:5061>
Contact: <sip:Anonymous@192.168.0.2:5060>
Call-ID: 3fd86a0228e349966aaf190a727b89e2@192.168.0.2:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.3
Date: Fri, 20 Apr 2012 21:56:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 497
v=0
o=root 1875648370 1875648370 IN IP4 192.168.0.2
s=Asterisk PBX 1.8.3
c=IN IP4 192.168.0.2
t=0 0
m=audio 12600 RTP/AVP 10 3 0 8 112 5 7 111 9 118 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:192.168.0.2:5061 --->
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK548003d0
From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as5430b682
Call-ID: 3fd86a0228e349966aaf190a727b89e2@192.168.0.2:5060
To: <sip:ivan@192.168.0.2:5061>
Contact: <sip:ivan@192.168.0.2:5061>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.0.2:5061 --->
SIP/2.0 180 Ringing
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK548003d0
User-Agent: Ekiga/3.2.7
From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as5430b682
Call-ID: 3fd86a0228e349966aaf190a727b89e2@192.168.0.2:5060
To: <sip:ivan@192.168.0.2:5061>;tag=14518c64-a189-e111-8257-000e7ba2889e
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:192.168.0.2:5061 --->
SIP/2.0 200 OK
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK548003d0
User-Agent: Ekiga/3.2.7
From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as5430b682
Call-ID: 3fd86a0228e349966aaf190a727b89e2@192.168.0.2:5060
To: <sip:ivan@192.168.0.2:5061>;tag=14518c64-a189-e111-8257-000e7ba2889e
Contact: <sip:ivan@192.168.0.2:5061>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 213
v=0
o=- 1334959006 1 IN IP4 192.168.0.2
s=Opal SIP Session
c=IN IP4 192.168.0.2
t=0 0
m=audio 5062 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
<------------->
--- (11 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.2:5062
list_route: hop: <sip:ivan@192.168.0.2:5061>
set_destination: Parsing <sip:ivan@192.168.0.2:5061> for address/port to send to
set_destination: set destination to 192.168.0.2:5061
Transmitting (no NAT) to 192.168.0.2:5061:
ACK sip:ivan@192.168.0.2:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK38d50219
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as5430b682
To: <sip:ivan@192.168.0.2:5061>;tag=14518c64-a189-e111-8257-000e7ba2889e
Contact: <sip:Anonymous@192.168.0.2:5060>
Call-ID: 3fd86a0228e349966aaf190a727b89e2@192.168.0.2:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.3
Content-Length: 0
---
> Channel SIP/ivan-00000001 was answered.
== Manager 'manager' logged off from 127.0.0.1
-- Executing [567@test:1] Answer("SIP/ivan-00000001", "") in new stack
-- Executing [567@test:2] NoOp("SIP/ivan-00000001", "") in new stack
-- Executing [567@test:3] SayUnixTime("SIP/ivan-00000001", "-1,UTC") in new stack
-- <SIP/ivan-00000001> Playing 'digits/day-3.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/mon-11.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/30.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/h-1.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/19.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/60.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/9.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/at.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/11.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/50.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/9.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/p-m.gsm' (language 'en')
-- Executing [567@test:4] SayUnixTime("SIP/ivan-00000001", ",CET,kMbdY") in new stack
-- <SIP/ivan-00000001> Playing 'digits/20.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/3.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/50.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/7.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/mon-3.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/h-20.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/2.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/thousand.gsm' (language 'en')
-- <SIP/ivan-00000001> Playing 'digits/12.gsm' (language 'en')
-- Executing [567@test:5] Playback("SIP/ivan-00000001", "vm-goodbye") in new stack
-- <SIP/ivan-00000001> Playing 'vm-goodbye.gsm' (language 'en')
-- Executing [567@test:6] Hangup("SIP/ivan-00000001", "") in new stack
== Spawn extension (test, 567, 6) exited non-zero on 'SIP/ivan-00000001'
Scheduling destruction of SIP dialog '3fd86a0228e349966aaf190a727b89e2@192.168.0.2:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:ivan@192.168.0.2:5061> for address/port to send to
set_destination: set destination to 192.168.0.2:5061
Reliably Transmitting (no NAT) to 192.168.0.2:5061:
BYE sip:ivan@192.168.0.2:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6071a5ce
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as5430b682
To: <sip:ivan@192.168.0.2:5061>;tag=14518c64-a189-e111-8257-000e7ba2889e
Call-ID: 3fd86a0228e349966aaf190a727b89e2@192.168.0.2:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.3
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:192.168.0.2:5061 --->
SIP/2.0 200 OK
CSeq: 103 BYE
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK6071a5ce
From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as5430b682
Call-ID: 3fd86a0228e349966aaf190a727b89e2@192.168.0.2:5060
To: <sip:ivan@192.168.0.2:5061>;tag=14518c64-a189-e111-8257-000e7ba2889e
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '3fd86a0228e349966aaf190a727b89e2@192.168.0.2:5060' Method: INVITE