<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:10pt">Hello,<br><br>It means that one of clients, is using 'silence suppression' mechanism
which sends audio frames that do not contain any samples.<br>Asterisk complains about silence supression and appears these warnings on CLI.<br>If the client turn off the silence suppression the message will disappear.<br><br>// Binan.<br><div style="font-family: times new roman, new york, times, serif; font-size: 10pt;"> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div dir="ltr"> <font size="2" face="Arial"> <hr size="1"> <b><span style="font-weight:bold;">Från:</span></b> Administrator TOOTAI <admin@tootai.net><br> <b><span style="font-weight: bold;">Till:</span></b> Asterisk-Users <asterisk-users@lists.digium.com> <br> <b><span style="font-weight: bold;">Skickat:</span></b> söndag, 21 oktober 2012 10:34<br> <b><span style="font-weight: bold;">Ämne:</span></b> [asterisk-users] Sound problem with format files but not codecs<br> </font> </div> <br>
Hi all,<br><br>on asterisk 1.8.16<br><br>[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<br>[2012-10-20 19:36:17] VERBOSE[743] res_musiconhold.c: -- Started music on hold, class 'TOOTAi', on Local/801@OFFICE-Numbers-e54a;2<br>[2012-10-20 19:36:17] WARNING[742] translate.c: no samples for ulawtolin<br>[2012-10-20 19:36:21] VERBOSE[742] pbx.c: == Spawn extension (from_to-OFFICE, 801, 23) exited non-zero on 'SIP/8081773619-00002f28'<br>[2012-10-20 19:36:21] VERBOSE[743] res_musiconhold.c: -- Stopped music on hold on Local/801@OFFICE-Numbers-e54a;2<br><br>or asterisk 10.8.0<br><br> -- Executing [801@macro-GeneralNumbers:1] Set("SIP/105-00000081", "CHANNEL(musicclass)=TOOTAi") in new stack<br> -- Executing [801@macro-GeneralNumbers:2] MusicOnHold("SIP/105-00000081", "") in new stack<br>
-- Started music on hold, class 'TOOTAi', on SIP/105-00000081<br>[2012-10-20 22:48:48] WARNING[22435]: translate.c:343 framein: no samples for g722tolin<br> -- Stopped music on hold on SIP/105-00000081<br><br>This is when calling extension:<br><br>exten=>801,1,Set(CHANNEL(musicclass)=TOOTAi)<br>exten=>801,n,MusicOnHold()<br>exten=>801,n,Hangup<br><br>What does mean those WARNINGS and how to solve this problem?<br><br>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.<br><br>Is this a bug? Did I forget something?<br><br>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<br><br>VERBOSE[19717] res_musiconhold.c:
-- Started music on hold, class 'default', on SIP/104-000000b3<br>VERBOSE[19717] res_musiconhold.c: -- Stopped music on hold on SIP/104-000000b3<br><br>which is MusicOnHold stop immediately.<br><br>On all servers wav files are installed, even try with original ones delivered with Asterisk.<br><br>Thanks for any hint<br><br>Regards<br>-- Daniel<br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> http://www.asterisk.org/hello<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br><br><br> </div> </div> </div></body></html>