Recording cisco call manager calls using Asterisk

Asterisk Asterisk Support
1

Dear Experts;

I am very new to asterisk.

I have successfully installed Asterisknow in my LAB.

We have a contact center and customer needs to record all the calls from or to the agents.

I have created a SIP trunk and recording profile as per the call manager 7.1 documentation and Installed Free Asterisk PBX in my VMWARe box and was able to browse the web .

My real concern is how to start the configuration in PBX side.

I got through a post we have to create sip.conf file and extension.conf in Asterisk.

Should we have to modify these files or have to create new files and where n how to start I am stuck.

Appreciate your help n feedback.

regards
Debashis Rout
CCIE VOICE #34888

2

There are two modes of using AsteriskNOW, GUI and non-GUI. Questions about the GUI mode should be directed to freepbx.org/forums/

If you are using it in non-GUI mode, you really need to go to asteriskdocs.org/ and have a good read.

3

Hi Experts;

My configuration is as below and the error coming in Asterisk is as below.

I am using Asterisknow CLI based.

----------------sip.conf---------------------------

[SIPTrk-cmpub]
disallow=all
host=IP-of-the-master
type=friend
context=from-trunk
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes

---------------------In extensions.conf -------------------

[from-trunk]

exten => 2380,1,Answer

exten => 2380,n,Noop( SIPCALLID ${SIPCALLID})

exten => 2380,n,Noop( UNIQUEID ${UNIQUEID})

exten => 2380,n,Noop( SIPHEADER From = ${SIP_HEADER(From)})

exten => 2380,n,Noop( SIPHEADER From = ${CUT(CUT(SIP_HEADER(From),;,7),>,1)})

exten => 2380,n,Set(remotedid=${CUT(CUT(SIP_HEADER(From),=,6),>,1)})

exten => 2380,n,Set(pseudodidi2=${CUT(SIP_HEADER(From),x-farendaddr,1)})

exten => 2380,n,Noop( ${remotedid})

exten => 2380,n,Record(Record${CALLERID(num)}${CUT(SIPCALLID,-,1)}-${CALLERID(num)}-${STRFTIME(${EPOCH},%Y%m%d-%H%M%S)}-${remotedid}-${CHANNEL:-2}%d:wav)

Asterisk-log–

[2013-09-04 00:43:18] VERBOSE[6626][C-00000001] pbx.c: – Executing [2380@from-sip-external:1] NoOp(“SIP/SIPTrk-cmpub-00000001”, "Received incoming SIP connection from

unknown peer to 2380") in new stack

[2013-09-04 00:43:18] VERBOSE[6626][C-00000001] pbx.c: – Executing [2380@from-sip-external:2] Set(“SIP/SIPTrk-cmpub-00000001”, “DID=2380”) in new stack

[2013-09-04 00:43:18] VERBOSE[6626][C-00000001] pbx.c: – Executing [2380@from-sip-external:3] Goto(“SIP/SIPTrk-cmpub-00000001”, “s,1”) in new stack

[2013-09-04 00:43:18] VERBOSE[6626][C-00000001] pbx.c: – Goto (from-sip-external,s,1)

[2013-09-04 00:43:18] VERBOSE[6626][C-00000001] pbx.c: – Executing [s@from-sip-external:1] GotoIf(“SIP/SIPTrk-cmpub-00000001”, “0?checklang:noanonymous”) in new stack

[2013-09-04 00:43:18] VERBOSE[6626][C-00000001] pbx.c: – Goto (from-sip-external,s,5)

[2013-09-04 00:43:18] VERBOSE[6626][C-00000001] pbx.c: – Executing [s@from-sip-external:5] Set(“SIP/SIPTrk-cmpub-00000001”, “TIMEOUT(absolute)=15”) in new stack

[2013-09-04 00:43:18] VERBOSE[6626][C-00000001] func_timeout.c: – Channel will hangup at 2013-09-04 00:43:33.843 AST.

[2013-09-04 00:43:18] VERBOSE[6626][C-00000001] pbx.c: – Executing [s@from-sip-external:6] Answer(“SIP/SIPTrk-cmpub-00000001”, “”) in new stack

[2013-09-04 00:43:19] VERBOSE[6625][C-00000000] pbx.c: – Executing [s@from-sip-external:7] Wait(“SIP/SIPTrk-cmpub-00000000”, “2”) in new stack

[2013-09-04 00:43:19] VERBOSE[6626][C-00000001] pbx.c: – Executing [s@from-sip-external:7] Wait(“SIP/SIPTrk-cmpub-00000001”, “2”) in new stack

[2013-09-04 00:43:21] VERBOSE[6625][C-00000000] pbx.c: – Executing [s@from-sip-external:8] Playback(“SIP/SIPTrk-cmpub-00000000”, “ss-noservice”) in new stack

[2013-09-04 00:43:21] VERBOSE[6625][C-00000000] file.c: – <SIP/SIPTrk-cmpub-00000000> Playing ‘ss-noservice.ulaw’ (language ‘en’)

[2013-09-04 00:43:21] VERBOSE[6626][C-00000001] pbx.c: – Executing [s@from-sip-external:8] Playback(“SIP/SIPTrk-cmpub-00000001”, “ss-noservice”) in new stack

[2013-09-04 00:43:21] VERBOSE[6626][C-00000001] file.c: – <SIP/SIPTrk-cmpub-00000001> Playing ‘ss-noservice.ulaw’ (language ‘en’)

[2013-09-04 00:43:26] VERBOSE[6625][C-00000000] pbx.c: – Executing [s@from-sip-external:9] PlayTones(“SIP/SIPTrk-cmpub-00000000”, “congestion”) in new stack

[2013-09-04 00:43:26] VERBOSE[6625][C-00000000] pbx.c: – Executing [s@from-sip-external:10] Congestion(“SIP/SIPTrk-cmpub-00000000”, “5”) in new stack

[2013-09-04 00:43:26] VERBOSE[6626][C-00000001] pbx.c: – Executing [s@from-sip-external:9] PlayTones(“SIP/SIPTrk-cmpub-00000001”, “congestion”) in new stack

[2013-09-04 00:43:26] VERBOSE[6626][C-00000001] pbx.c: – Executing [s@from-sip-external:10] Congestion(“SIP/SIPTrk-cmpub-00000001”, “5”) in new stack

[2013-09-04 00:43:28] VERBOSE[6625][C-00000000] pbx.c: == Spawn extension (from-sip-external, s, 10) exited non-zero on ‘SIP/SIPTrk-cmpub-00000000’

[2013-09-04 00:43:28] VERBOSE[6626][C-00000001] pbx.c: == Spawn extension (from-sip-external, s, 10) exited non-zero on ‘SIP/SIPTrk-cmpub-00000001’

[2013-09-04 00:43:28] VERBOSE[6625][C-00000000] pbx.c: – Executing [h@from-sip-external:1] Hangup(“SIP/SIPTrk-cmpub-00000000”, “”) in new stack

[2013-09-04 00:43:28] VERBOSE[6626][C-00000001] pbx.c: – Executing [h@from-sip-external:1] Hangup(“SIP/SIPTrk-cmpub-00000001”, “”) in new stack

[2013-09-04 00:43:28] VERBOSE[6626][C-00000001] pbx.c: == Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/SIPTrk-cmpub-00000001’

[2013-09-04 00:43:28] VERBOSE[6625][C-00000000] pbx.c: == Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/SIPTrk-cmpub-00000000’

Appreciate your suggestion.

Regards
debashis

4

Your dialplan fragment is from a different context from the one that is actually being used. Again, if you are using the FreePBX GUI, that GUI is not supported here.