What are your actual file names (/etc/asterisk/musiconhold/Frederic Chopin Polonaised Op. 40-2.wav?)
_____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Saul Sent: Wednesday, September 16, 2009 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold That was a good shot in the dark, but sadly renaming it to something simple (and removing all non ascii in the process) does not correct this. On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas <da...@debsinc.com> wrote: Just a shot in the dark but could MOH be choking on the long file names? (does it work on fred_chopin_pol_1)? _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Saul Sent: Wednesday, September 16, 2009 4:18 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1 These files were generated by SoX: Channels : 1 Sample Rate : 8000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM Endian Type : little Reverse Nibbles: no Reverse Bits : no Comment : 'Processed by SoX' This prints in the asterisk console when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rvvvvv): == Using SIP RTP CoS mark 5 -- Executing [xxxx...@phones:1] Goto("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "1xxxxxxxxxx,1") in new stack -- Goto (phones,1xxxxxxxxxx,1) -- Executing [1xxxxxxx...@phones:1] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "oldcidnum=0") in new stack -- Executing [1xxxxxxx...@phones:2] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "CALLERID(name)=""") in new stack -- Executing [1xxxxxxx...@phones:3] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "CALLERID(num)=xxxxxxxxxx") in new stack -- Executing [1xxxxxxx...@phones:4] Monitor("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "wav,/tmp/out 0 2009-09-17 03h 04m 51s CST xxxxxxxxxx,m") in new stack -- Executing [1xxxxxxx...@phones:5] Gosub("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "ExternalDial,s,1(1xxxxxxxxxx)") in new stack -- Executing [...@externaldial:1] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "LOCAL(num)=1xxxxxxxxxx") in new stack -- Executing [...@externaldial:2] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "~~EXTEN~~=s") in new stack -- Executing [...@externaldial:3] Dial("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "SIP/1xxxxxxx...@link2voip-sw1,120") in new stack == Using SIP RTP CoS mark 5 -- Called 1xxxxxxx...@link2voip-sw1 -- SIP/link2voip-sw1-02477668 is making progress passing it to SIP/ATA-xxxxxxxxxx-L1-024b6d88 -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xxxxxxxxxx-L1-024b6d88 -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 > doing dnsmgr_lookup for 'sip.ca2.link2voip.com' > doing dnsmgr_lookup for 'sip.ca1.link2voip.com' == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xxxxxxxxxx-L1-024b6d88' Any thoughts or ideas? If there were an error I could work on solving that, but there is none... Thanks. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
