Re: [asterisk-users] Sound problem with format files but not codecs
Le 22/10/2012 04:27, Binan AL Halabi a écrit : Hello, It means that one of clients, is using 'silence suppression' mechanism which sends audio frames that do not contain any samples. Asterisk complains about silence supression and appears these warnings on CLI. If the client turn off the silence suppression the message will disappear. Hi Binan, silence suppression is already turned off Regards // Binan. *Från:* Administrator TOOTAI *Till:* Asterisk-Users *Skickat:* söndag, 21 oktober 2012 10:34 *Ämne:* [asterisk-users] Sound problem with format files but not codecs Hi all, on asterisk 1.8.16 [2012-10-20 19:36:17] VERBOSE[743] pbx.c:-- Executing [801@OFFICE-Numbers:2] MusicOnHold("Local/801@OFFICE-Numbers-e54a;2", "") in new stack [2012-10-20 19:36:17] VERBOSE[743] res_musiconhold.c:-- Started music on hold, class 'TOOTAi', on Local/801@OFFICE-Numbers-e54a;2 [2012-10-20 19:36:17] WARNING[742] translate.c: no samples for ulawtolin [2012-10-20 19:36:21] VERBOSE[742] pbx.c: == Spawn extension (from_to-OFFICE, 801, 23) exited non-zero on 'SIP/8081773619-2f28' [2012-10-20 19:36:21] VERBOSE[743] res_musiconhold.c:-- Stopped music on hold on Local/801@OFFICE-Numbers-e54a;2 or asterisk 10.8.0 -- Executing [801@macro-GeneralNumbers:1] Set("SIP/105-0081", "CHANNEL(musicclass)=TOOTAi") in new stack -- Executing [801@macro-GeneralNumbers:2] MusicOnHold("SIP/105-0081", "") in new stack -- Started music on hold, class 'TOOTAi', on SIP/105-0081 [2012-10-20 22:48:48] WARNING[22435]: translate.c:343 framein: no samples for g722tolin -- Stopped music on hold on SIP/105-0081 This is when calling extension: exten=>801,1,Set(CHANNEL(musicclass)=TOOTAi) exten=>801,n,MusicOnHold() exten=>801,n,Hangup What does mean those WARNINGS and how to solve this problem? MeetMe, Voicemail or holding a call are working fine. From what I understand, codecs are used in channels and format for handling files. In both cases, two different servers, asterisk is compiled from tar.gz and in menuselect all codecs and formats are activated. Is this a bug? Did I forget something? On a third server I run latest Elastix with an asterisk 1.8.16 version. On this server I have no MusicOnHold at all even during calls. Logs show VERBOSE[19717] res_musiconhold.c:-- Started music on hold, class 'default', on SIP/104-00b3 VERBOSE[19717] res_musiconhold.c:-- Stopped music on hold on SIP/104-00b3 which is MusicOnHold stop immediately. On all servers wav files are installed, even try with original ones delivered with Asterisk. Thanks for any hint -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound problem with format files but not codecs
Hello, It means that one of clients, is using 'silence suppression' mechanism which sends audio frames that do not contain any samples. Asterisk complains about silence supression and appears these warnings on CLI. If the client turn off the silence suppression the message will disappear. // Binan. Från: Administrator TOOTAI Till: Asterisk-Users Skickat: söndag, 21 oktober 2012 10:34 Ämne: [asterisk-users] Sound problem with format files but not codecs Hi all, on asterisk 1.8.16 [2012-10-20 19:36:17] VERBOSE[743] pbx.c: -- Executing [801@OFFICE-Numbers:2] MusicOnHold("Local/801@OFFICE-Numbers-e54a;2", "") in new stack [2012-10-20 19:36:17] VERBOSE[743] res_musiconhold.c: -- Started music on hold, class 'TOOTAi', on Local/801@OFFICE-Numbers-e54a;2 [2012-10-20 19:36:17] WARNING[742] translate.c: no samples for ulawtolin [2012-10-20 19:36:21] VERBOSE[742] pbx.c: == Spawn extension (from_to-OFFICE, 801, 23) exited non-zero on 'SIP/8081773619-2f28' [2012-10-20 19:36:21] VERBOSE[743] res_musiconhold.c: -- Stopped music on hold on Local/801@OFFICE-Numbers-e54a;2 or asterisk 10.8.0 -- Executing [801@macro-GeneralNumbers:1] Set("SIP/105-0081", "CHANNEL(musicclass)=TOOTAi") in new stack -- Executing [801@macro-GeneralNumbers:2] MusicOnHold("SIP/105-0081", "") in new stack -- Started music on hold, class 'TOOTAi', on SIP/105-0081 [2012-10-20 22:48:48] WARNING[22435]: translate.c:343 framein: no samples for g722tolin -- Stopped music on hold on SIP/105-0081 This is when calling extension: exten=>801,1,Set(CHANNEL(musicclass)=TOOTAi) exten=>801,n,MusicOnHold() exten=>801,n,Hangup What does mean those WARNINGS and how to solve this problem? MeetMe, Voicemail or holding a call are working fine. From what I understand, codecs are used in channels and format for handling files. In both cases, two different servers, asterisk is compiled from tar.gz and in menuselect all codecs and formats are activated. Is this a bug? Did I forget something? On a third server I run latest Elastix with an asterisk 1.8.16 version. On this server I have no MusicOnHold at all even during calls. Logs show VERBOSE[19717] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/104-00b3 VERBOSE[19717] res_musiconhold.c: -- Stopped music on hold on SIP/104-00b3 which is MusicOnHold stop immediately. On all servers wav files are installed, even try with original ones delivered with Asterisk. Thanks for any hint Regards -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sound problem with format files but not codecs
Hi all, on asterisk 1.8.16 [2012-10-20 19:36:17] VERBOSE[743] pbx.c: -- Executing [801@OFFICE-Numbers:2] MusicOnHold("Local/801@OFFICE-Numbers-e54a;2", "") in new stack [2012-10-20 19:36:17] VERBOSE[743] res_musiconhold.c: -- Started music on hold, class 'TOOTAi', on Local/801@OFFICE-Numbers-e54a;2 [2012-10-20 19:36:17] WARNING[742] translate.c: no samples for ulawtolin [2012-10-20 19:36:21] VERBOSE[742] pbx.c: == Spawn extension (from_to-OFFICE, 801, 23) exited non-zero on 'SIP/8081773619-2f28' [2012-10-20 19:36:21] VERBOSE[743] res_musiconhold.c: -- Stopped music on hold on Local/801@OFFICE-Numbers-e54a;2 or asterisk 10.8.0 -- Executing [801@macro-GeneralNumbers:1] Set("SIP/105-0081", "CHANNEL(musicclass)=TOOTAi") in new stack -- Executing [801@macro-GeneralNumbers:2] MusicOnHold("SIP/105-0081", "") in new stack -- Started music on hold, class 'TOOTAi', on SIP/105-0081 [2012-10-20 22:48:48] WARNING[22435]: translate.c:343 framein: no samples for g722tolin -- Stopped music on hold on SIP/105-0081 This is when calling extension: exten=>801,1,Set(CHANNEL(musicclass)=TOOTAi) exten=>801,n,MusicOnHold() exten=>801,n,Hangup What does mean those WARNINGS and how to solve this problem? MeetMe, Voicemail or holding a call are working fine. From what I understand, codecs are used in channels and format for handling files. In both cases, two different servers, asterisk is compiled from tar.gz and in menuselect all codecs and formats are activated. Is this a bug? Did I forget something? On a third server I run latest Elastix with an asterisk 1.8.16 version. On this server I have no MusicOnHold at all even during calls. Logs show VERBOSE[19717] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/104-00b3 VERBOSE[19717] res_musiconhold.c: -- Stopped music on hold on SIP/104-00b3 which is MusicOnHold stop immediately. On all servers wav files are installed, even try with original ones delivered with Asterisk. Thanks for any hint Regards -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users