Hi people, I have a problem with Music On Hold, it is stopped just after starting... This is the log: [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [s at test:1] NoOp("SIP/7PBX-08229d18", "Start") in new stack [Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold' [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [s at test:2] MusicOnHold("SIP/7PBX-08229d18", "") in new stack [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format slin [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Started music on hold, class 'default', on channel 'SIP/7PBX-08229d18' [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 160 sample intervals [Jun 17 19:20:22] DEBUG[20784] channel.c: Auto-deactivating generator [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format ulaw [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, Anahi Ludue?a _________________________________________________________________ S? el protagonista de GQ con Messenger y Vodafone Blackberry. ?Y gana premios! http://serviciosmoviles.es.msn.com/messenger/vodafone.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100617/fe5b230c/attachment.htm
I see that moh is trying sln format, then ulaw, then failing. Do you have moh files in either of these formats? _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anahi Ludue?a Sent: Thursday, June 17, 2010 2:24 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Music on Hold problema Hi people, I have a problem with Music On Hold, it is stopped just after starting... This is the log: [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [s at test:1] NoOp("SIP/7PBX-08229d18", "Start") in new stack [Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold' [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [s at test:2] MusicOnHold("SIP/7PBX-08229d18", "") in new stack [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format slin [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Started music on hold, class 'default', on channel 'SIP/7PBX-08229d18' [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 160 sample intervals [Jun 17 19:20:22] DEBUG[20784] channel.c: Auto-deactivating generator [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format ulaw [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, _____ Anahi Ludue?a _____ Disfruta de Hotmail y Messenger en tu m?vil con YOIGO. ?Hazlo <http://serviciosmoviles.es.msn.com/hotmail/yoigo.aspx> ya! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100617/1baf71d3/attachment.htm
Hi, As Danny said, asterisk is looking for slin or ulaw files. Are your wav files in any of these formats? Did you just copied them from somewhere without changing their format? Also note they should be 8KHz mono 16 bit files. You can do this in a simple utility like Windows Recorder. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-18 10:35 AM, "Danny Nicholas" <danny at debsinc.com> wrote: Post the /var/lib/asterisk/mohmp3 listing and musiconhold.conf ------------------------------ *From:* asterisk-users-bounces at lists.digium.com [mailto: asterisk-users-bounces at lists.digium.com] *On Behalf Of *Anahi Ludue?a *Sent:* Friday, June 18, 2010 9:18 AM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Music on Hold problema Any ideas, please? ________________________________ Anahi Ludue?a ____________________... -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100618/ba6b844e/attachment.htm
The moh conf file seems good. It is the standard implementation and should have worked. Just wondering if your end devices, whether they are IP phones or softphones, are setup to listen to some different codecs than ulaw and slin? Or in your sip.conf when declaring extensions you are not putting the correct codecs in the 'allow=' declaration. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-23 10:33 AM, "Anahi Ludue?a" <a_luduena at hotmail.com> wrote: Please, I need help with this... ________________________________ Anahi Ludue?a ________________________________ From: a_... Date: Fri, 18 Jun 2010 15:12:25 +0000 Subject: Re: [asterisk-users] Music on Hold problema The list of /var/lib/asterisk/mohmp3 is: -rw... -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100623/cb674a06/attachment.htm