Call recordings not working if pin set enbled

raloheni · February 26, 2010, 4:49pm

hi
i am using asterisknow with freepbx2.7 .the problem i am facing is
Call recording is not working if i enable the pinset option.
for long distance call i have enbled the pinset by dialling zero infront of number.

But call recording is working for local calls for which the pinset is not enbled .

raloheni · February 27, 2010, 5:56pm

hi please check the log taken without sip debug ext 413 destination 098918198xx
test3*CLI>
– Executing [[email protected]:1] Macro(“SIP/413-00000013”, “user-callerid|SKIPTTL|”) in new stack
– Executing [[email protected]:1] Set(“SIP/413-00000013”, “AMPUSER=413”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/413-00000013”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/413-00000013”, “1|Set|REALCALLERIDNUM=413”) in new stack
– Executing [[email protected]:4] Set(“SIP/413-00000013”, “AMPUSER=413”) in new stack
– Executing [[email protected]:5] Set(“SIP/413-00000013”, “AMPUSERCIDNAME=413”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/413-00000013”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/413-00000013”, “AMPUSERCID=413”) in new stack
– Executing [[email protected]:8] Set(“SIP/413-00000013”, “CALLERID(all)=“413” <413>”) in new stack
– Executing [[email protected]:9] ExecIf(“SIP/413-00000013”, “0|Set|CHANNEL(language)=”) in new stack

test3*CLI>
– Executing [[email protected]:10] GotoIf(“SIP/413-00000013”, “1?continue”) in new stack

test3*CLI>
– Goto (macro-user-callerid,s,19)

test3*CLI>
– Executing [[email protected]:19] NoOp(“SIP/413-00000013”, “Using CallerID “413” <413>”) in new stack

test3*CLI>
– Executing [[email protected]:2] Macro(“SIP/413-00000013”, “pinsets|1|1”) in new stack

test3*CLI>
– Executing [[email protected]:1] GotoIf(“SIP/413-00000013”, “1 = 1?cdr|1”) in new stack

test3*CLI>
– Goto (macro-pinsets,cdr,1)

test3*CLI>
– Executing [[email protected]:1] ExecIf(“SIP/413-00000013”, “1|Authenticate|/etc/asterisk/pinset_1|a”) in new stack

test3*CLI>
– <SIP/413-00000013> Playing ‘agent-pass’ (language ‘en’)

test3*CLI>
== Parsing ‘/etc/asterisk/manager.conf’: Found

test3*CLI>
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found

test3*CLI>
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found

test3*CLI>
== Manager ‘admin’ logged on from 127.0.0.1

test3*CLI>
== Manager ‘admin’ logged off from 127.0.0.1

test3*CLI>
– <SIP/413-00000013> Playing ‘auth-thankyou’ (language ‘en’)

test3*CLI>
– Executing [[email protected]cro-pinsets:2] ExecIf(“SIP/413-00000013”, “1|ResetCDR|”) in new stack
– Executing [[email protected]:3] Set(“SIP/413-00000013”, “_NODEST=”) in new stack

test3*CLI>
– Executing [[email protected]:4] Macro(“SIP/413-00000013”, “record-enable|413|OUT|”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/413-00000013”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/413-00000013”, “recordingcheck|20100227-230745|1267292261.37”) in new stack

test3*CLI>
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck

test3*CLI>
recordingcheck|20100227-230745|1267292261.37: Outbound recording enabled.

test3*CLI>
recordingcheck|20100227-230745|1267292261.37: CALLFILENAME=OUT413-20100227-230745-1267292261.37

test3*CLI>
– AGI Script recordingcheck completed, returning 0

test3*CLI>
– Executing [[email protected]:999] MixMonitor(“SIP/413-00000013”, “OUT413-20100227-230745-1267292261.37.wav||”) in new stack

test3*CLI>
== Begin MixMonitor Recording SIP/413-00000013

test3*CLI>
– Executing [[email protected]:5] Macro(“SIP/413-00000013”, “dialout-trunk|1|09891819857||”) in new stack

test3*CLI>
– Executing [[email protected]:1] Set(“SIP/413-00000013”, “DIAL_TRUNK=1”) in new stack

test3*CLI>
– Executing [[email protected]:2] GosubIf(“SIP/413-00000013”, “0?sub-pincheck|s|1”) in new stack

test3*CLI>
– Executing [[email protected]:3] GotoIf(“SIP/413-00000013”, “0?disabletrunk|1”) in new stack

test3*CLI>
– Executing [[email protected]:4] Set(“SIP/413-00000013”, “DIAL_NUMBER=09891819857”) in new stack

test3*CLI>
– Executing [[email protected]:5] Set(“SIP/413-00000013”, “DIAL_TRUNK_OPTIONS=tr”) in new stack

test3*CLI>
– Executing [[email protected]:6] Set(“SIP/413-00000013”, “OUTBOUND_GROUP=OUT_1”) in new stack

test3*CLI>
– Executing [[email protected]:7] GotoIf(“SIP/413-00000013”, “1?nomax”) in new stack

test3*CLI>
– Goto (macro-dialout-trunk,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/413-00000013”, “0?skipoutcid”) in new stack

test3*CLI>
– Executing [[email protected]:10] Set(“SIP/413-00000013”, “DIAL_TRUNK_OPTIONS=”) in new stack

test3*CLI>
– Executing [[email protected]:11] Macro(“SIP/413-00000013”, “outbound-callerid|1”) in new stack

test3*CLI>
– Executing [[email protected]:1] ExecIf(“SIP/413-00000013”, “0|SetCallerPres|”) in new stack

test3*CLI>
– Executing [[email protected]:2] ExecIf(“SIP/413-00000013”, “0|Set|REALCALLERIDNUM=413”) in new stack

test3*CLI>
– Executing [[email protected]:3] GotoIf(“SIP/413-00000013”, “1?normcid”) in new stack

test3*CLI>
– Goto (macro-outbound-callerid,s,6)
– Executing [[email protected]:6] Set(“SIP/413-00000013”, “USEROUTCID=”) in new stack

test3*CLI>
– Executing [[email protected]:7] Set(“SIP/413-00000013”, “EMERGENCYCID=”) in new stack

test3*CLI>
– Executing [[email protected]:8] Set(“SIP/413-00000013”, “TRUNKOUTCID=”) in new stack

test3*CLI>
– Executing [[email protected]:9] GotoIf(“SIP/413-00000013”, “1?trunkcid”) in new stack

test3*CLI>
– Goto (macro-outbound-callerid,s,12)

test3*CLI>
– Executing [[email protected]:12] ExecIf(“SIP/413-00000013”, “0|Set|CALLERID(all)=”) in new stack

test3*CLI>
– Executing [[email protected]:13] ExecIf(“SIP/413-00000013”, “0|Set|CALLERID(all)=”) in new stack

test3*CLI>
– Executing [[email protected]:14] ExecIf(“SIP/413-00000013”, “0|SetCallerPres|prohib_passed_screen”) in new stack

test3*CLI>
– Executing [[email protected]:12] ExecIf(“SIP/413-00000013”, “1|AGI|fixlocalprefix”) in new stack

test3*CLI>
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix

test3*CLI>
> fixlocalprefix: Using pattern XXXXXXXXXX

test3*CLI>
> fixlocalprefix: Using pattern XXXXXXXX

test3*CLI>
> fixlocalprefix: Using pattern XXX

test3*CLI>
> fixlocalprefix: Using pattern 0Z.

test3*CLI>
== fixlocalprefix: Dialpattern 0Z. matched. 09891819857 -> 09891819857

test3*CLI>
– AGI Script fixlocalprefix completed, returning 0

test3*CLI>
– Executing [[email protected]:13] Set(“SIP/413-00000013”, “OUTNUM=09891819857”) in new stack

test3*CLI>
– Executing [[email protected]:14] Set(“SIP/413-00000013”, “custom=DAHDI/g0”) in new stack

test3*CLI>
– Executing [[email protected]:15] ExecIf(“SIP/413-00000013”, “0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)”) in new stack

test3*CLI>
– Executing [[email protected]:16] Macro(“SIP/413-00000013”, “dialout-trunk-predial-hook|”) in new stack

test3*CLI>
– Executing [[email protected]:1] MacroExit(“SIP/413-00000013”, “”) in new stack

test3*CLI>
– Executing [[email protected]:17] GotoIf(“SIP/413-00000013”, “0?bypass|1”) in new stack

test3*CLI>
– Executing [[email protected]:18] GotoIf(“SIP/413-00000013”, “0?customtrunk”) in new stack

test3*CLI>
– Executing [[email protected]:19] Dial(“SIP/413-00000013”, “DAHDI/g0/09891819857|300|”) in new stack

test3*CLI>
– Requested transfer capability: 0x00 - SPEECH

test3*CLI>
– Called g0/09891819857

test3*CLI>
– DAHDI/32-1 is proceeding passing it to SIP/413-00000013

test3*CLI>
== Parsing ‘/etc/asterisk/manager.conf’: Found

test3*CLI>
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found

test3*CLI>
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found

test3*CLI>
== Manager ‘admin’ logged on from 127.0.0.1

test3*CLI>
== Manager ‘admin’ logged off from 127.0.0.1

test3*CLI>
– DAHDI/32-1 is ringing

test3*CLI>
– DAHDI/32-1 answered SIP/413-00000013

test3*CLI>
== Parsing ‘/etc/asterisk/manager.conf’: Found

test3*CLI>
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found

test3*CLI>
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found

test3*CLI>
== Manager ‘admin’ logged on from 127.0.0.1

test3*CLI>
== Manager ‘admin’ logged off from 127.0.0.1

test3*CLI>
– Executing [[email protected]:1] Macro(“SIP/413-00000013”, “hangupcall|”) in new stack

test3*CLI>
– Executing [[email protected]:1] GotoIf(“SIP/413-00000013”, “1?skiprg”) in new stack

test3*CLI>
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/413-00000013”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)

test3*CLI>
– Executing [[email protected]:7] GotoIf(“SIP/413-00000013”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)

test3*CLI>
– Executing [[email protected]:9] Hangup(“SIP/413-00000013”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/413-00000013’ in macro ‘hangupcall’
== Spawn h extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/413-00000013’

test3*CLI>
– Hungup ‘DAHDI/32-1’

test3*CLI>
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/413-00000013’ in macro ‘dialout-trunk’

test3*CLI>
== Spawn extension (from-internal, 09891819857, 5) exited non-zero on ‘SIP/413-00000013’

test3*CLI>
== MixMonitor close filestream
== End MixMonitor Recording SIP/413-00000013

test3*CLI>
== Parsing ‘/etc/asterisk/manager.conf’: Found

test3*CLI>
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found

test3*CLI>
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found

test3*CLI>
== Manager ‘admin’ logged on from 127.0.0.1

test3*CLI>
== Manager ‘admin’ logged off from 127.0.0.1

test3*CLI> exit

Executing last minute cleanups

plindheimer · February 27, 2010, 7:01am

post a CLI output of the call going out the pinset route that is suppose to be recording and is not please.

raloheni · February 27, 2010, 8:05am

hi heres my cli output
ext is 413 and destination is 0994012xxxx

test3*CLI>
== Parsing ‘/etc/asterisk/manager.conf’: Found

test3*CLI>
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1

test3*CLI>
== Manager ‘admin’ logged off from 127.0.0.1

test3*CLI>

<— SIP read from 182.67.200.64:5060 —>
INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 182.67.200.64:5060;branch=z9hG4bKfd295a06f42d895dcf16f4377cf75a2f

From: 413 sip:[email protected];tag=63f4e861

To: 0994012xxxx sip:[email protected]

Call-ID: [email protected]

CSeq: 1 INVITE

Contact: sip:[email protected]

Max-Forwards: 70

User-agent: ET747

Content-Type: application/sdp

Content-Length: 279

v=0

o=- 1992602 1992602 IN IP4 182.67.200.64

s=SIP Session

c=IN IP4 182.67.200.64

t=0 0

m=audio 10004 RTP/AVP 4 0 8 18 56

a=rtpmap:4 G723/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=ptime:60

a=rtpmap:56 telephone-event/8000

a=fmtp:56 0-15

<------------->

test3*CLI>
— (11 headers 13 lines) —

test3*CLI>
Sending to 182.67.200.64 : 5060 (no NAT)

test3*CLI>
Using INVITE request as basis request - [email protected]

test3*CLI>

<— Reliably Transmitting (NAT) to 182.67.200.64:5060 —>
SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 182.67.200.64:5060;branch=z9hG4bKfd295a06f42d895dcf16f4377cf75a2f;received=182.67.200.64

From: 413 sip:[email protected];tag=63f4e861

To: 0994012xxxx sip:[email protected];tag=as7827826d

Call-ID: [email protected]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“7d46ef3f”

Content-Length: 0

<------------>

test3*CLI>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)

test3*CLI>
Found user ‘413’

test3*CLI>

<— SIP read from 182.67.200.64:5060 —>
ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 182.67.200.64:5060;branch=z9hG4bKfd295a06f42d895dcf16f4377cf75a2f

From: 413 sip:[email protected];tag=63f4e861

To: 0994012xxxx sip:[email protected];tag=as7827826d

Call-ID: [email protected]

CSeq: 1 ACK

Max-Forwards: 70

User-agent: ET747

Content-Length: 0

<------------->

test3*CLI>
— (9 headers 0 lines) —

test3*CLI>

<— SIP read from 182.67.200.64:5060 —>
INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 182.67.200.64:5060;branch=z9hG4bKda7a812473b530359d9c0f40edb1835c

From: 413 sip:[email protected];tag=63f4e861

To: 0994012xxxx sip:[email protected]

Call-ID: [email protected]

CSeq: 2 INVITE

Contact: sip:[email protected]

Max-Forwards: 70

Proxy-Authorization: Digest algorithm=MD5,nonce=“7d46ef3f”,realm=“asterisk”,response=“004632481149c855b7f8bed63d02c028”,uri=“sip:[email protected]:5060”,username=“413”

User-agent: ET747

Content-Type: application/sdp

Content-Length: 279

v=0

o=- 1992602 1992602 IN IP4 182.67.200.64

s=SIP Session

c=IN IP4 182.67.200.64

t=0 0

m=audio 10004 RTP/AVP 4 0 8 18 56

a=rtpmap:4 G723/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=ptime:60

a=rtpmap:56 telephone-event/8000

a=fmtp:56 0-15

<------------->

test3*CLI>
— (12 headers 13 lines) —

test3*CLI>
Sending to 182.67.200.64 : 5060 (NAT)

test3*CLI>
Using INVITE request as basis request - [email protected]

test3*CLI>
Found user ‘413’

test3*CLI>
Found RTP audio format 4

test3*CLI>
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18

test3*CLI>
Found RTP audio format 56

test3*CLI>
Found audio description format G723 for ID 4

test3*CLI>
Found audio description format PCMU for ID 0

test3*CLI>
Found audio description format PCMA for ID 8

test3*CLI>
Found audio description format G729 for ID 18

test3*CLI>
Found audio description format telephone-event for ID 56

test3*CLI>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

test3*CLI>
Peer audio RTP is at port 182.67.200.64:10004

test3*CLI>
Looking for 0994012xxxx in from-internal (domain 122.165.10.61)

test3*CLI>
list_route: hop: sip:[email protected]

test3*CLI>

<— Transmitting (NAT) to 182.67.200.64:5060 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 182.67.200.64:5060;branch=z9hG4bKda7a812473b530359d9c0f40edb1835c;received=182.67.200.64

From: 413 sip:[email protected];tag=63f4e861

To: 0994012xxxx sip:[email protected]

Call-ID: [email protected]

CSeq: 2 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Contact: sip:[email protected]

Content-Length: 0

<------------>

test3*CLI>
– Executing [[email protected]:1] Macro(“SIP/413-00000007”, “user-callerid|SKIPTTL|”) in new stack

test3*CLI>
– Executing [[email protected]:1] Set(“SIP/413-00000007”, “AMPUSER=413”) in new stack

test3*CLI>
– Executing [[email protected]:2] GotoIf(“SIP/413-00000007”, “0?report”) in new stack

test3*CLI>
– Executing [[email protected]:3] ExecIf(“SIP/413-00000007”, “1|Set|REALCALLERIDNUM=413”) in new stack

test3*CLI>
– Executing [[email protected]:4] Set(“SIP/413-00000007”, “AMPUSER=413”) in new stack

test3*CLI>
– Executing [[email protected]:5] Set(“SIP/413-00000007”, “AMPUSERCIDNAME=413”) in new stack

test3*CLI>
– Executing [[email protected]:6] GotoIf(“SIP/413-00000007”, “0?report”) in new stack

test3*CLI>
– Executing [[email protected]:7] Set(“SIP/413-00000007”, “AMPUSERCID=413”) in new stack

test3*CLI>
– Executing [[email protected]:8] Set(“SIP/413-00000007”, “CALLERID(all)=“413” <413>”) in new stack

test3*CLI>
– Executing [[email protected]:9] ExecIf(“SIP/413-00000007”, “0|Set|CHANNEL(language)=”) in new stack

test3*CLI>
– Executing [[email protected]:10] GotoIf(“SIP/413-00000007”, “1?continue”) in new stack

test3*CLI>
– Goto (macro-user-callerid,s,19)

test3*CLI>
– Executing [[email protected]:19] NoOp(“SIP/413-00000007”, “Using CallerID “413” <413>”) in new stack

test3*CLI>
– Executing [[email protected]:2] Macro(“SIP/413-00000007”, “pinsets|1|0”) in new stack

test3*CLI>
– Executing [[email protected]:1] GotoIf(“SIP/413-00000007”, “0 = 1?cdr|1”) in new stack

test3*CLI>
– Executing [[email protected]:2] ExecIf(“SIP/413-00000007”, “1|Authenticate|/etc/asterisk/pinset_1”) in new stack

test3*CLI>
Audio is at 182.67.200.195 port 18530

test3*CLI>
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

test3*CLI>

<— Reliably Transmitting (NAT) to 182.67.200.64:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 182.67.200.64:5060;branch=z9hG4bKda7a812473b530359d9c0f40edb1835c;received=182.67.200.64

From: 413 sip:[email protected];tag=63f4e861

To: 0994012xxxx sip:[email protected];tag=as605c1ddc

Call-ID: [email protected]

CSeq: 2 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Contact: sip:[email protected]

Content-Type: application/sdp

Content-Length: 263

v=0

o=root 3265 3265 IN IP4 182.67.200.195

s=session

c=IN IP4 182.67.200.195

t=0 0

m=audio 18530 RTP/AVP 0 8 56

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:56 telephone-event/8000

a=fmtp:56 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

<------------>

test3*CLI>
– <SIP/413-00000007> Playing ‘agent-pass’ (language ‘en’)

test3*CLI>

<— SIP read from 182.67.200.64:5060 —>
ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 182.67.200.64:5060;branch=z9hG4bK640834c650b4220f4942b0089d534052

From: sip:[email protected];tag=63f4e861

To: sip:[email protected];tag=as605c1ddc

Call-ID: [email protected]

CSeq: 2 ACK

Max-Forwards: 70

User-agent: ET747

Content-Length: 0

raloheni · February 27, 2010, 8:10am