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@mydomain.com set(skip_record_check=true)** **2013-04-29 13:07:51.694194 [DEBUG] mod_dptools.c:1344 sofia/internal/208@mydomain.com SET [skip_record_check]=[true]** **EXECUTE sofia/internal/208@mydomain.com set(skip_urgent_check=true)** **2013-04-29 13:07:51.694194 [DEBUG] mod_dptools.c:1344 sofia/internal/208@mydomain.com SET [skip_urgent_check]=[true]** *EXECUTE sofia/internal/208@mydomain.com answer() 2013-04-29 13:07:51.704184 [DEBUG] sofia_glue.c:3351 AUDIO RTP [sofia/internal/208@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@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@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@mydomain.com [BREAK] 2013-04-29 13:07:51.704184 [DEBUG] switch_channel.c:3395 (sofia/internal/208@mydomain.com) Callstate Change RINGING -> ACTIVE 2013-04-29 13:07:51.704184 [NOTICE] mod_dptools.c:1176 Channel [sofia/internal/208@mydomain.com] has been answered 2013-04-29 13:07:51.704184 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/208@mydomain.com [BREAK] 2013-04-29 13:07:51.704184 [DEBUG] sofia.c:5574 Channel sofia/internal/208@mydomain.com entering state [completed][200] *EXECUTE sofia/internal/208@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@mydomain.com [BREAK] 2013-04-29 13:07:51.724183 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/208@mydomain.com [BREAK] 2013-04-29 13:07:51.724183 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/208@mydomain.com [BREAK] 2013-04-29 13:07:51.734185 [DEBUG] sofia.c:5574 Channel sofia/internal/208@mydomain.com entering state [ready][200] 2013-04-29 13:07:51.734185 [DEBUG] switch_ivr.c:192 Codec Activated L16@8000hz 1 channels 20ms 2013-04-29 13:07:51.844184 [DEBUG] switch_ivr_play_say.c:1338 Codec Activated L16@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@mydomain.com Push codec L16:70 2013-04-29 13:08:07.264182 [DEBUG] switch_core_codec.c:244 sofia/internal/208@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@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@telefaks.de <mailto:lists@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@telefaks.de> <mailto:lists@telefaks.de> wrote:
Anybody has a clue how to solve this?
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
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