[Freeswitch-users] mod_voicemail: skip instructions after recording

Michael Collins msc at freeswitch.org
Mon Apr 29 23:09:29 MSD 2013 

Peter,

FYI, I tested this patch and it worked for me. I suspect that you need to
use export instead of set for the variables. For testing I simply used the
Local_Extension in the default.xml file. I added these two lines directly
before the voicemail action, so it looks like this:

        <action application="export" data="skip_record_urgent_check=true"/>
        <action application="export" data="skip_record_check=true"/>
        <action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>

The message gets recorded and then the caller simply hears "goodbye" and
gets disconnected.
-MC



On Mon, Apr 29, 2013 at 8:18 AM, Michael Collins <msc at freeswitch.org> wrote:

> Thanks for testing. If I get a moment to work on this today I will. It's
> probably something simple.
> -MC
>
>
> On Mon, Apr 29, 2013 at 4:15 AM, Peter Steinbach <lists at telefaks.de>wrote:
>
>>  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.com200)
>> *
>> 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>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> <lists at telefaks.de> wrote:
>>>
>>>
>>>  Anybody has a clue how to solve this?
>>>
>>>  --
>>> Brian Westbrian at freeswitch.org
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>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
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>>> With kind regards
>>> Peter Steinbach
>>>
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>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
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>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
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>>
>>
>> --
>> With kind regards
>> Peter Steinbach
>>
>> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
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>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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