[asterisk-users] Music on Hold

[Dan Saul]

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 [xxxxxxx at phones:1] Goto("SIP/ATA-xxxxxxxxxx-L1-024b6d88",
"1xxxxxxxxxx,1") in new stack
    -- Goto (phones,1xxxxxxxxxx,1)
    -- Executing [1xxxxxxxxxx at phones:1]
MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "oldcidnum=0") in new stack
    -- Executing [1xxxxxxxxxx at phones:2]
MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "CALLERID(name)=""") in new stack
    -- Executing [1xxxxxxxxxx at phones:3]
MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "CALLERID(num)=xxxxxxxxxx") in new
stack
    -- Executing [1xxxxxxxxxx at 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 [1xxxxxxxxxx at phones:5]
Gosub("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "ExternalDial,s,1(1xxxxxxxxxx)") in
new stack
    -- Executing [s at ExternalDial:1] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88",
"LOCAL(num)=1xxxxxxxxxx") in new stack
    -- Executing [s at ExternalDial:2] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88",
"~~EXTEN~~=s") in new stack
    -- Executing [s at ExternalDial:3] Dial("SIP/ATA-xxxxxxxxxx-L1-024b6d88",
"SIP/1xxxxxxxxxx at link2voip-sw1,120") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 1xxxxxxxxxx at 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.
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