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call recorder asterisk with callmanager 8.6.2

Hello,

I have a cucm 8.6 and the client want record calls with asterisk.

I have configured the callmanager with I have read in documentation, I mean, I configured the trunk sip, record profile , route pattern and the appication user and  I have associated the telephones wiht built in bridge activated.

I see the sip traffic between Callmanager and traffic. I see the invite with xnearend and far end. But the Asterisk declined thsi sip and don't record the call.

Do you know if the Asterisk support this method for record calls? ,I mean the method that the callmanager uses with the record profiles

Thanks in advance

Regards

Raul

13 REPLIES
New Member

call recorder asterisk with callmanager 8.6.2

Hello,

it works with 8.5, so it should work with 8.6 too.

Have you configured something on astersik side ?

One easy test is to call the recorder extension from a cisco phone. I guess you'll hear Astersik auto answer message (if you install a ready to use distribution).

I also suggest to log on Astersik machine and see asterisk console.

Regards,

NH

New Member

call recorder asterisk with callmanager 8.6.2

Hello,

Could you find any method to record Cisco UCM calls to Asterisk. If you have a solution can you share please ?

Regards,

Tolga

New Member

call recorder asterisk with callmanager 8.6.2

Hello,

where are you stucked ?

You need to configure a trunk on CCM and one (or more if you have slaves) on the Asterisk box. For instance this is my SIP conf on Asterisk :

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

[SIPTrk-cmsub01]
disallow=all
host=IP-of-the-slave

type=friend
context=from-trunk
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes

In extensions.conf add this :

[from-trunk]

exten => 98123,1,Answer

exten => 98123,n,Noop( SIPCALLID  ${SIPCALLID})

exten => 98123,n,Noop( UNIQUEID ${UNIQUEID})

exten => 98123,n,Noop( SIPHEADER From = _${SIP_HEADER(From)}_)

exten => 98123,n,Noop( SIPHEADER From = _${CUT(CUT(SIP_HEADER(From),\;,7),>,1)}_)

exten => 98123,n,Set(remotedid=${CUT(CUT(SIP_HEADER(From),=,6),>,1)})

exten => 98123,n,Set(pseudodidi2=${CUT(SIP_HEADER(From),x-farendaddr,1)})

exten => 98123,n,Noop( ${remotedid})

exten => 98123,n,Record(Record${CALLERID(num)}_${CUT(SIPCALLID,-,1)}_-${CALLERID(num)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${remotedid}-${CHANNEL:-2}%d:wav)

you can add an Hangup if you want.

In the record profile, just mention that the target extension is 98123.

Each time a user will call or receive call on his hardware phone (not soft phone like Jabber), two calls will be established with Asterisk (one in and one out). After the call you'll need to merge the two recorded files. I put one stream at the left and one at the right.

Don't forget to activate the bridge on CCM.

Hope it will help.

New Member

call recorder asterisk with callmanager 8.6.2

Hello,

Thank for your reply,

I have done CUCM Trunk Setup,

Activated bridge on CUCM.

I can see call logs but there is no recording.

Do you have any idea about this case.

Regards

New Member

call recorder asterisk with callmanager 8.6.2

So you see incoming call in asterisk ?

Have you cut and paste the extensions setup ? Reload the conf on Asterisk ?

If so, you should see all the different "calls" on the console. Is it complaining when reaching Record step ?

Just drop the 10 lasts line from console.

New Member

call recorder asterisk with callmanager 8.6.2

Hi again,

Thanks for your help, I had set up extensions.conf.

The log file is pasted below.

Any idea ?

Regards,

The log file says;

2013-06-05 08:40:11] VERBOSE[1840] pbx.c: -- Executing [29999@from-trunk:1] Answer("SIP/SIPTrk-cmpub-00000000", "") in new stack

[2013-06-05 08:40:12] VERBOSE[1713] netsock2.c: == Using SIP RTP TOS bits 184

[2013-06-05 08:40:12] VERBOSE[1713] netsock2.c: == Using SIP RTP CoS mark 5

[2013-06-05 08:40:12] VERBOSE[1841] pbx.c: -- Executing [29999@from-trunk:1] Answer("SIP/SIPTrk-cmpub-00000001", "") in new stack

[2013-06-05 08:40:12] VERBOSE[1840] pbx.c: -- Executing [29999@from-trunk:2] NoOp("SIP/SIPTrk-cmpub-00000000", " SIPCALLID 62257500-1ae1cf3b-2e5f-a00000a@10.0.0.10") in new stack

[2013-06-05 08:40:12] VERBOSE[1840] pbx.c: -- Executing [29999@from-trunk:3] NoOp("SIP/SIPTrk-cmpub-00000000", " UNIQUEID 1370410811.0") in new stack

[2013-06-05 08:40:12] VERBOSE[1840] pbx.c: -- Executing [29999@from-trunk:4] NoOp("SIP/SIPTrk-cmpub-00000000", " SIPHEADER From = _"CC Agent" <1000>;tag=24182~556927dd-159e-4492-9f47-ffb30ba7dd72-20615157_") in new stack

[2013-06-05 08:40:12] VERBOSE[1840] pbx.c: -- Executing [29999@from-trunk:5] NoOp("SIP/SIPTrk-cmpub-00000000", " SIPHEADER From = _x-farendaddr=05418412230_") in new stack

[2013-06-05 08:40:12] VERBOSE[1840] pbx.c: -- Executing [29999@from-trunk:6] Set("SIP/SIPTrk-cmpub-00000000", "remotedid=05418412230") in new stack

[2013-06-05 08:40:12] VERBOSE[1840] pbx.c: -- Executing [29999@from-trunk:7] Set("SIP/SIPTrk-cmpub-00000000", "pseudodidi2="CC Agent" <1000>

[2013-06-05 08:40:12] VERBOSE[1840] pbx.c: -- Executing [29999@from-trunk:8] NoOp("SIP/SIPTrk-cmpub-00000000", "05418412230") in new stack

[2013-06-05 08:40:12] VERBOSE[1840] pbx.c: -- Executing [29999@from-trunk:9] Record("SIP/SIPTrk-cmpub-00000000", "Record1000_62257500_-1000-20130605-084012-05418412230-00%d:wav") in new stack

[2013-06-05 08:40:12] VERBOSE[1840] file.c: -- Playing 'beep.ulaw' (language 'en')

[2013-06-05 08:40:12] VERBOSE[1841] pbx.c: -- Executing [29999@from-trunk:2] NoOp("SIP/SIPTrk-cmpub-00000001", " SIPCALLID 62be0b80-1ae1cf3c-2e60-a00000a@10.0.0.10") in new stack

[2013-06-05 08:40:12] VERBOSE[1841] pbx.c: -- Executing [29999@from-trunk:3] NoOp("SIP/SIPTrk-cmpub-00000001", " UNIQUEID 1370410812.1") in new stack

[2013-06-05 08:40:12] VERBOSE[1841] pbx.c: -- Executing [29999@from-trunk:4] NoOp("SIP/SIPTrk-cmpub-00000001", " SIPHEADER From = _"CC Agent" <1000>;tag=24183~556927dd-159e-4492-9f47-ffb30ba7dd72-20615161_") in new stack

[2013-06-05 08:40:12] VERBOSE[1841] pbx.c: -- Executing [29999@from-trunk:5] NoOp("SIP/SIPTrk-cmpub-00000001", " SIPHEADER From = _x-farendaddr=05418412230_") in new stack

[2013-06-05 08:40:12] VERBOSE[1841] pbx.c: -- Executing [29999@from-trunk:6] Set("SIP/SIPTrk-cmpub-00000001", "remotedid=05418412230") in new stack

[2013-06-05 08:40:12] VERBOSE[1841] pbx.c: -- Executing [29999@from-trunk:7] Set("SIP/SIPTrk-cmpub-00000001", "pseudodidi2="CC Agent" <1000>

[2013-06-05 08:40:12] VERBOSE[1841] pbx.c: -- Executing [29999@from-trunk:8] NoOp("SIP/SIPTrk-cmpub-00000001", "05418412230") in new stack

[2013-06-05 08:40:12] VERBOSE[1841] pbx.c: -- Executing [29999@from-trunk:9] Record("SIP/SIPTrk-cmpub-00000001", "Record1000_62be0b80_-1000-20130605-084012-05418412230-01%d:wav") in new stack

New Member

call recorder asterisk with callmanager 8.6.2

Hi,

I have send the asterisk log. Still not recording.

Do you have any idea ?

Thanks

New Member

call recorder asterisk with callmanager 8.6.2

Hum... Stupid question, but how do you know that there's no records ?

Have you got a look to /var/asterisk/... something ?

It is the directory where all the records are stored. I think you should find some files like Record1000*.wav

New Member

call recorder asterisk with callmanager 8.6.2

Hi dear;

I have the same configuration in call manager and also asterisk as above.

I am using call manager 7.0 and asterisknow freePBX.

The call was able to reach Asterisk but there is no recording happening.

The logs as below..

Appreciate your great help to resolve this issue.

[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: -- 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: -- 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'

Regards

Debashis

New Member

call recorder asterisk with callmanager 8.6.2

Hi,  Nicolas Horchower

"After the call you'll need to merge the two recorded files"

How to merge the two recorded files to one.

Thanks a lot.

New Member

call recorder asterisk with callmanager 8.6.2

Please, Can any one give us more details.

Thanks

New Member

call recorder asterisk with callmanager 8.6.2

Hello,

this is a "simple" script to do the merge. It is not 100% perfect, some files will need to be merged manually :

create a folder /var/lib/asterisk/sounds/bad/ to store not matched files then :

------------------------

for NAME in $(find /var/lib/asterisk/sounds/  -maxdepth 1 -type f -name "Record*" | cut -d "/" -f6)

do

        echo Proceed $NAME

        in=${NAME%_*}

        if [ -e $NAME ]

        then

                out=${NAME#N*-}

                soxmix $in*.wav MergedRecord-$out

                if [ "$?" -eq "0" ]

                then

                        rm -f $in*.wav

                else

                        mv $in*.wav bad

                fi

#       else

        #       echo                   $NAME NOT FOUND

        fi

#echo -------------------------------

done

------------------------

some optimizations :

-run this after the call, and correctly handle the two "legs" of the call to not move "bad" calls.

-create a web interface to browse the records.

Regards,

NH

New Member

call recorder asterisk with callmanager 8.6.2

Thanks a lot Nicolas Horchower

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