Call recordings not working if pin set enbled

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 .

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

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

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>
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)

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

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>
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)

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

please turn off you sip debug traces and re-post so we can see the dialplan and not have to sift through all the sip packets which are not relevant for this. Thanks.

(btw - I tested your scenario on 2.7 with a pinset and call recording was engaged, fyi, but get a clean trace and maybe someone can see what is going on for you)

HI
as per the cli log its displaying outbound recording is enbled and its also showing the uniqueid and file name with .wav ext.

and also i checked /var/spool/asterisk/monitor that file is their
but in callmonitor window its not displaying the play or download option.

but its displaying play and download option for non pinset calls.

well then it sounds like the issue is a different issue, call recording is happening, just ARI isn’t showing the call.

Assuming there is not a ticket open, you may want to file a bug on that but please provide plenty of detailed info for a working and non-working case such as:

  • Call File Name generated
  • UNIQUE ID stored in the CDR record for the call

thanks a lot