[Freeswitch-users] mod_voicemail: skip instructions after recording
Peter Steinbach
lists at telefaks.de
Mon Apr 29 15:15:58 MSD 2013
Hello Michael,
thanks for your quick support.
I applied the patch and tested it, however, it did not suppress the
messages:
Here's an excerpt from the log with the relevant parts marked in bold:
*EXECUTE sofia/internal/208 at mydomain.com set(skip_record_check=true)**
**2013-04-29 13:07:51.694194 [DEBUG] mod_dptools.c:1344
sofia/internal/208 at mydomain.com SET [skip_record_check]=[true]**
**EXECUTE sofia/internal/208 at mydomain.com set(skip_urgent_check=true)**
**2013-04-29 13:07:51.694194 [DEBUG] mod_dptools.c:1344
sofia/internal/208 at mydomain.com SET [skip_urgent_check]=[true]**
*EXECUTE sofia/internal/208 at mydomain.com answer()
2013-04-29 13:07:51.704184 [DEBUG] sofia_glue.c:3351 AUDIO RTP
[sofia/internal/208 at mydomain.com] 192.168.178.220 port 12522 ->
192.168.178.118 port 3000 codec: 8 ms: 20
2013-04-29 13:07:51.704184 [DEBUG] switch_rtp.c:1975 Starting timer
[soft] 160 bytes per 20ms
2013-04-29 13:07:51.704184 [DEBUG] switch_rtp.c:2331 RTCP send rate is:
5000 and packet rate is: 20000 Remote Port: 3001
2013-04-29 13:07:51.704184 [DEBUG] switch_rtp.c:1342 Setting RTCP remote
addr to 192.168.178.118:3001
2013-04-29 13:07:51.704184 [DEBUG] sofia_glue.c:3615 Set 2833 dtmf send
payload to 101
2013-04-29 13:07:51.704184 [DEBUG] sofia_glue.c:3621 Set 2833 dtmf
receive payload to 101
2013-04-29 13:07:51.704184 [DEBUG] sofia_glue.c:3648
sofia/internal/208 at mydomain.com Set rtp dtmf delay to 40
2013-04-29 13:07:51.704184 [DEBUG] mod_sofia.c:856 Local SDP
sofia/internal/208 at mydomain.com:
v=0
o=FreeSWITCH 1367221149 1367221150 IN IP4 192.168.178.220
s=FreeSWITCH
c=IN IP4 192.168.178.220
t=0 0
m=audio 12522 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
2013-04-29 13:07:51.704184 [DEBUG] switch_core_session.c:830 Send signal
sofia/internal/208 at mydomain.com [BREAK]
2013-04-29 13:07:51.704184 [DEBUG] switch_channel.c:3395
(sofia/internal/208 at mydomain.com) Callstate Change RINGING -> ACTIVE
2013-04-29 13:07:51.704184 [NOTICE] mod_dptools.c:1176 Channel
[sofia/internal/208 at mydomain.com] has been answered
2013-04-29 13:07:51.704184 [DEBUG] switch_core_session.c:975 Send signal
sofia/internal/208 at mydomain.com [BREAK]
2013-04-29 13:07:51.704184 [DEBUG] sofia.c:5574 Channel
sofia/internal/208 at mydomain.com entering state [completed][200]
*EXECUTE sofia/internal/208 at mydomain.com voicemail(default mydomain.com
200)*
2013-04-29 13:07:51.724183 [DEBUG] switch_core_session.c:975 Send signal
sofia/internal/208 at mydomain.com [BREAK]
2013-04-29 13:07:51.724183 [DEBUG] switch_core_session.c:975 Send signal
sofia/internal/208 at mydomain.com [BREAK]
2013-04-29 13:07:51.724183 [DEBUG] switch_core_session.c:975 Send signal
sofia/internal/208 at mydomain.com [BREAK]
2013-04-29 13:07:51.734185 [DEBUG] sofia.c:5574 Channel
sofia/internal/208 at mydomain.com entering state [ready][200]
2013-04-29 13:07:51.734185 [DEBUG] switch_ivr.c:192 Codec Activated
L16 at 8000hz 1 channels 20ms
2013-04-29 13:07:51.844184 [DEBUG] switch_ivr_play_say.c:1338 Codec
Activated L16 at 8000hz 1 channels 20ms
2013-04-29 13:07:51.884186 [DEBUG] switch_rtp.c:3694 Correct ip/port
confirmed.
2013-04-29 13:07:58.204183 [DEBUG] switch_ivr_play_say.c:1721 done
playing file
/usr/local/freeswitch/storage/voicemail/default/mydomain.com/200/greeting_1.wav
2013-04-29 13:07:59.224199 [DEBUG] switch_ivr_play_say.c:595 Raw Codec
Activated
2013-04-29 13:07:59.224199 [DEBUG] switch_core_codec.c:219
sofia/internal/208 at mydomain.com Push codec L16:70
2013-04-29 13:08:07.264182 [DEBUG] switch_core_codec.c:244
sofia/internal/208 at mydomain.com Restore previous codec PCMA:8.
2013-04-29 13:08:07.264182 [DEBUG] switch_ivr_play_say.c:70 No language
specified - Using [en]
2013-04-29 13:08:07.264182 [WARNING] switch_xml.c:2356 Invalid UTF-8
character to ampersand, skip it
2013-04-29 13:08:07.264182 [WARNING] switch_xml.c:2356 Invalid UTF-8
character to ampersand, skip it
2013-04-29 13:08:07.264182 [WARNING] switch_xml.c:2356 Invalid UTF-8
character to ampersand, skip it
2013-04-29 13:08:07.264182 [WARNING] switch_xml.c:2356 Invalid UTF-8
character to ampersand, skip it
*2013-04-29 13:08:07.264182 [DEBUG] switch_ivr_play_say.c:247 Handle
play-file:[voicemail/vm-press.wav] (en:en)**
*2013-04-29 13:08:07.264182 [DEBUG] switch_ivr_play_say.c:1338 Codec
Activated L16 at 8000hz 1 channels 20ms
2013-04-29 13:08:07.664181 [DEBUG] switch_ivr_play_say.c:1721 done
playing file
/usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav
Best regards
Peter
On 04/29/13 07:15, Michael Collins wrote:
> Peter,
>
> After looking at mod_voicemail.c I could tell that there's no way to
> do this currently. However, like Brian said, there isn't a whole lot
> to it. Here's a simple patch for you to test.
>
> To use this just set the channel variables 'skip_record_check' and
> 'skip_urgent_check' to true. It should then just disconnect the
> channel once the recording is done. The first variable turns off the
> record check ("1 to listen, 2 to save, 3 to re-record") and the second
> turns off the urgent check ("to mark this message urgent, press 1...").
>
> Also, if this seems to work then please open a new jira of type
> "improvement" for project "FreeSWITCH" and attach this patch. (If you
> make any adjustments, please create a new patch and attach it instead
> of this file.)
>
> Let us know how it goes.
> -Michael
>
>
>
> On Sun, Apr 28, 2013 at 3:30 AM, Peter Steinbach <lists at telefaks.de
> <mailto:lists at telefaks.de>> wrote:
>
> Hello Brian,
>
> I was thinking about that too. But is there a way of having a
> solution without patching the code?
>
>
> Best regards
> Peter
>
>
>
>
>
> On 04/26/13 15:45, Brian West wrote:
>> Should be fairly simple to find where that ends in the code and add the proper bits so it doesn't play that.
>>
>> Its patches just like what you described that got me into coding C back when I started with Asterisk.
>>
>> On Apr 26, 2013, at 7:01 AM, Peter Steinbach <lists at telefaks.de> <mailto:lists at telefaks.de> wrote:
>>
>>> Anybody has a clue how to solve this?
>> --
>> Brian West
>> brian at freeswitch.org <mailto:brian at freeswitch.org>
>> FreeSWITCH Solutions, LLC
>> PO BOX PO BOX 2531
>> Brookfield, WI 53008-2531
>> Twitter: @FreeSWITCH_Wire
>> T: +1.918.420.9266 <tel:%2B1.918.420.9266> | F: +1.918.420.9267 <tel:%2B1.918.420.9267> | M: +1.918.424.WEST
>> iNUM: +883 5100 1420 9266 <tel:%2B883%205100%201420%209266>
>> ISN: 410*543
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
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>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
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>>
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>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de <http://telefaks.de>
> Internet: www.telefaks.de <http://www.telefaks.de>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
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> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
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>
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>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
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> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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--
With kind regards
Peter Steinbach
Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de
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