[Freeswitch-users] mod_voicemail: skip instructions after recording

Mon Apr 29 19:18:48 MSD 2013

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
>>
>> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
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>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
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>
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>
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>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
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>

-- 
Michael S Collins
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