mod_voicemail: skip instructions after recording
Hello, I am wondering how I can skip the instructions to the caller AFTER recoding his message. Skipping the instructions BEFORE recording works nicely, but I have found no information how to avoid the instructions being played AFTER the caller has spoken his message. I simply want mod_voicemail to hangup as soon as he has detected that the recording has finished. Anybody has a clue how to solve this? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
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> wrote:
Anybody has a clue how to solve this?
-- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543
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> wrote:
Anybody has a clue how to solve this?
-- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
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> 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> <lists@telefaks.de> wrote:
Anybody has a clue how to solve this?
-- Brian Westbrian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
-- With kind regards Peter Steinbach
Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de Internet: www.telefaks.de
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
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?
-- Brian West brian@freeswitch.org <mailto:brian@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- 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: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
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@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@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.com200) * 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>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> <lists@telefaks.de> wrote:
Anybody has a clue how to solve this?
-- Brian Westbrian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
-- With kind regards Peter Steinbach
Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de Internet: www.telefaks.de
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
-- With kind regards Peter Steinbach
Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de Internet: www.telefaks.de
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
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@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@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@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.com200) * 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>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> <lists@telefaks.de> wrote:
Anybody has a clue how to solve this?
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Hello Michael, I changed to export and entered: <action application="export" data="skip_record_check=true"/> <action application="export" data="skip_urgent_check=true"/> <action application="export" data="skip_record_urgent_check=true"/> <action application="answer"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} 200"/> The behaviour however is still the same. It still asks "press 1 for ....." Looking at the channel variables I can see that they are prefixed with "variable_" variable_skip_record_check true variable_skip_record_urgent_check true variable_skip_urgent_check true variable_export_vars skip_record_check,skip_urgent_check,skip_record_urgent_check And in the code they are queried without "variable_". Maybe the problem is there? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
Maybe you didn't actually fully apply the patch and re-install the module. Usually when it makes no sense it means you are not even changing anything *shrug just a guess* On Mon, Apr 29, 2013 at 4:22 PM, Peter Steinbach <lists@telefaks.de> wrote:
Hello Michael,
I changed to export and entered: <action application="export" data="skip_record_check=true"/> <action application="export" data="skip_urgent_check=true"/> <action application="export" data="skip_record_urgent_check=true"/> <action application="answer"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} 200"/>
The behaviour however is still the same. It still asks "press 1 for ....."
Looking at the channel variables I can see that they are prefixed with "variable_" variable_skip_record_check true variable_skip_record_urgent_check true variable_skip_urgent_check true variable_export_vars skip_record_check,skip_urgent_check,skip_record_urgent_check
And in the code they are queried without "variable_".
Maybe the problem is there?
-- With kind regards Peter Steinbach
Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
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participants (4)
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Anthony Minessale
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Brian West
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Michael Collins
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Peter Steinbach