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RECORDING OPTION *55

rrichiez
Posts: 243
 Member Since:
2006-12-07
Submitted by rrichiez on Mon, 07/23/2007 - 11:05am.

Is there plans to make this feature work. I really need this option. or at least if im doing it wrong some doc to le me know how to. when im on the phone and want to record that conversation i hit *55 and it should start recording but it does not. only if i go to the ivr setting of the extension and in recordings select all the time instead of demand it works. please has anyone got it to work . i really urgently need this function but if i leave it on to record always it will record everything and that will fill up my box with crap. please help anyone.

rrichiez

‹ Broken Ring PSTN Calling: "All Circuits Are Busy" ›


kerryg
Posts: 5533
 Member Since:
2006-05-31
Record on demand is *1 by
Mon, 07/23/2007 - 11:17am

Record on demand is *1 by default

--

Kerry Garrison
trixbox Community Director
http://www.trixbox.com/support-and-training/training
http://L7Foto.com / http://AsteriskTutorials.com / http://voipspeak.net



rrichiez
Posts: 243
 Member Since:
2006-12-07
*55
Mon, 07/23/2007 - 11:23am

Yes Kerry

Thank you for your quick response but by default it is on demand as you stated, but when i make or revieve a call im under the impresion that if you want to record that call you should hit " *55" during that call and it should start recording. In my case it does not. Am I doing anything wrong?



kerryg
Posts: 5533
 Member Since:
2006-05-31
Did you change it from the
Mon, 07/23/2007 - 11:38am

Did you change it from the default of *1?

--

Kerry Garrison
trixbox Community Director
http://www.trixbox.com/support-and-training/training
http://L7Foto.com / http://AsteriskTutorials.com / http://voipspeak.net



w5waf
Posts: 711
 Member Since:
2006-06-09
Try punching *1 after making
Mon, 07/23/2007 - 11:38am

Try punching *1 after making sure the "automon" in features.conf is not commented out.

--

Bill Ford - FtOCC
City of Vicksburg
www.vicksburg.org



rrichiez
Posts: 243
 Member Since:
2006-12-07
*55
Mon, 07/23/2007 - 12:04pm

in the features.conf i have this

and its set to *1

automon => *1

Kerry on the IVR recording settings its set to
Record INCOMING: On-Demand

what should I change?



rrichiez
Posts: 243
 Member Since:
2006-12-07
*55
Mon, 07/23/2007 - 12:04pm

in the features.conf i have this

and its set to *1

automon => *1

Kerry on the IV recording settings its set to
Record INCOMING: On-Demand

what should I change?



kerryg
Posts: 5533
 Member Since:
2006-05-31
Make sure that for the
Mon, 07/23/2007 - 12:17pm

Make sure that for the extension, you have on-demand enabled and then dial *1 during a call. I have no clue where you get *55 from, that is the default for disallow intercom.

--

Kerry Garrison
trixbox Community Director
http://www.trixbox.com/support-and-training/training
http://L7Foto.com / http://AsteriskTutorials.com / http://voipspeak.net



rrichiez
Posts: 243
 Member Since:
2006-12-07
*55
Mon, 07/23/2007 - 1:04pm

Kerry

any way i try *1 and it doesnt work . when i go to the call monitor i see no recording Thanks
Rrichiez



mario
Posts: 45
 Member Since:
2007-03-13
put this in features.conf
Mon, 07/23/2007 - 2:06pm

put this in features.conf under general section

featuredigittimeout=1500;

and then amportal stop and amportal start from cli

and retry !!



excessnet
Posts: 5
 Member Since:
2007-07-28
internal to internal ?
Sun, 07/29/2007 - 10:57am

Are you calling internal to internal ? Maybe it's becose of that ?

It's not working for me too btw...



aragh0rn
Posts: 6
 Member Since:
2007-01-31
internal to internal ?
Fri, 09/14/2007 - 9:26pm

Is any diference ?

Im have in feature.conf:

[general]

featuredigittimeout = 1500

[featuremap]
blindxfer => ## ; Blind Transfer
disconnect => ** ; Disconnect Call
automon => *1 ; One Touch Record
;atxfer => *2 ; Attended Xfer

Also i have in call setings wW for bouth, in/out
And also I have enabel in the extension Record on demand.

But it does not work.

Asterisk 1.4.11
FreePBX 2.3.0.3



rrichiez
Posts: 243
 Member Since:
2006-12-07
what are you using!
Fri, 09/14/2007 - 10:24pm

I have 3 ip phones they all work and i also have an analog phone this one does not work. i guess that if you are using analog phones its not working it didnt work for me until i realized this.

rrichiez



aragh0rn
Posts: 6
 Member Since:
2007-01-31
I proved whith
Sat, 09/15/2007 - 6:19am

I proved whith ip-phones(GXP-2020), soft-phones(ekiga), analog-phones. Al gave me the same result.
When I press *1 it hang-up.

asterisk report:

############
== Spawn extension (macro-dial, s, 10) exited non-zero on
'SIP/501-08b1ea00' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'SIP/501-08b1ea00' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/501-08b1ea00'
-- Executing [h@macro-dial:1] Macro("SIP/501-08b1ea00",
"hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/501-08b1ea00",
"w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/501-08b1ea00", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/501-08b1ea00",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/501-08b1ea00",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/501-08b1ea00",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/501-08b1ea00", "")
in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/501-08b1ea00' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/501-08b1ea00'

############

Thanks



aragh0rn
Posts: 6
 Member Since:
2007-01-31
SOLVED
Sun, 09/16/2007 - 5:57pm

It was a problem whith the "courtesy tone". thats why it hanged up.

when i increase the output, asterisk report:

asterisk1*CLI> -- SIP/501-093c8028 is ringing
-- SIP/501-093c8028 is ringing
asterisk1*CLI> -- SIP/501-093c8028 answered SIP/30-093c1e98
-- SIP/501-093c8028 answered SIP/30-093c1e98
asterisk1*CLI> [Sep 16 21:03:17] DTMF[12667]: channel.c:2346 __ast_read: DTMF end '*' received on SIP/30-093c1e98, duration 480 ms
[Sep 16 21:03:17] DTMF[12667]: channel.c:2382 __ast_read: DTMF begin emulation of '*' with duration 480 queued on SIP/30-093c1e98
[Sep 16 21:03:17] DTMF[12667]: channel.c:2346 __ast_read: DTMF end '*' received on SIP/30-093c1e98, duration 480 ms
[Sep 16 21:03:17] DTMF[12667]: channel.c:2382 __ast_read: DTMF begin emulation of '*' with duration 480 queued on SIP/30-093c1e98
asterisk1*CLI> [Sep 16 21:03:17] DTMF[12667]: channel.c:2346 __ast_read: DTMF end '1' received on SIP/30-093c1e98, duration 640 ms
[Sep 16 21:03:17] DTMF[12667]: channel.c:2352 __ast_read: DTMF end '1' put into dtmf queue on SIP/30-093c1e98
[Sep 16 21:03:17] DTMF[12667]: channel.c:2346 __ast_read: DTMF end '1' received on SIP/30-093c1e98, duration 640 ms
[Sep 16 21:03:17] DTMF[12667]: channel.c:2352 __ast_read: DTMF end '1' put into dtmf queue on SIP/30-093c1e98
asterisk1*CLI> [Sep 16 21:03:17] DTMF[12667]: channel.c:2465 __ast_read: DTMF end emulation of '*' queued on SIP/30-093c1e98
[Sep 16 21:03:17] DTMF[12667]: channel.c:2465 __ast_read: DTMF end emulation of '*' queued on SIP/30-093c1e98
asterisk1*CLI> [Sep 16 21:03:17] DTMF[12667]: channel.c:2215 __ast_read: DTMF begin emulation of '1' with duration 100 queued on SIP/30-093c1e98
[Sep 16 21:03:17] DTMF[12667]: channel.c:2215 __ast_read: DTMF begin emulation of '1' with duration 100 queued on SIP/30-093c1e98
asterisk1*CLI> [Sep 16 21:03:17] DTMF[12667]: channel.c:2465 __ast_read: DTMF end emulation of '1' queued on SIP/30-093c1e98
[Sep 16 21:03:17] WARNING[12667]: file.c:563 ast_openstream_full: File beep does not exist in any format
[Sep 16 21:03:17] WARNING[12667]: file.c:813 ast_streamfile: Unable to open beep (format 0x4 (ulaw)): No such file or directory
[Sep 16 21:03:17] WARNING[12667]: res_features.c:555 builtin_automonitor: Failed to play courtesy tone!
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/30-093c1e98' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/30-093c1e98' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/30-093c1e98'
-- Executing [h@macro-dial:1] Macro("SIP/30-093c1e98", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/30-093c1e98", "w") in new stack
[Sep 16 21:03:17] DTMF[12667]: channel.c:2465 __ast_read: DTMF end emulation of '1' queued on SIP/30-093c1e98
[Sep 16 21:03:17] WARNING[12667]: file.c:563 ast_openstream_full: File beep does not exist in any format
[Sep 16 21:03:17] WARNING[12667]: file.c:813 ast_streamfile: Unable to open beep (format 0x4 (ulaw)): No such file or directory
[Sep 16 21:03:17] WARNING[12667]: res_features.c:555 builtin_automonitor: Failed to play courtesy tone!
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/30-093c1e98' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/30-093c1e98' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/30-093c1e98'
-- Executing [h@macro-dial:1] Macro("SIP/30-093c1e98", "hangupcall") in new stack

i didn't have the beep.alaw !!!!

Thanks for all. Now all work as it should



soulofmischief87
Posts: 1
 Member Since:
2008-02-22
hmmm
Fri, 05/09/2008 - 1:16pm

i downloaded the beep.alaw into /var/lib/asterisk/sounds but the calls hang up after i press * even before i get to the 1 i have wW and i have features.conf im using g729 do i need a g729 "courtesy tone".
[general]

featuredigittimeout = 1500

[featuremap]
blindxfer => ## ; Blind Transfer
disconnect => ** ; Disconnect Call
automon => *1 ; One Touch Record
;atxfer => *2 ; Attended Xfer



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