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 <[email protected]> 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:* [email protected] [mailto: > [email protected]] *On Behalf Of *Dan Saul > *Sent:* Wednesday, September 16, 2009 4:18 PM > *To:* [email protected] > *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
