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Dan Saul

2009-Sep-16 21:17 UTC

[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/Fr?d?ric Chopin - Polonaises Op. 40-2
    File: /etc/asterisk/musiconhold/Fr?d?ric 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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Danny Nicholas

2009-Sep-16 21:37 UTC

head link

[asterisk-users] Music on Hold

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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Saul
Sent: Wednesday, September 16, 2009 4:18 PM
To: asterisk-users at 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/Fr?d?ric Chopin - Polonaises Op. 40-2
    File: /etc/asterisk/musiconhold/Fr?d?ric 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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Miguel Molina

2009-Sep-16 21:54 UTC

head link

[asterisk-users] Music on Hold

Dan Saul escribi?:> 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/Fr?d?ric Chopin - Polonaises Op. 40-2
>     File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1I would use a more friendly filename. That special accents and spaces 
maybe are confusing asterisk when it tries to read the files. Try 
renaming to chopin_op40-1 and chopin_op40-2 for example.

Cheers,

-- 
Ing. Miguel Molina
Grupo de Tecnolog?a
Millenium Phone Center

Cyprus VoIP

2009-Sep-29 11:31 UTC

head link

[asterisk-users] Music On Hold

Hello,

We need help in debugging Music On Hold on our Asterisk 1.6.1.6

 From the SIP debug, I see that an extension sends an INVITE of the call 
to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but 
I don't see in the console any reference to the call being placed on hold.

When I typed "moh show files", I see the wav files of the 
/var/lib/asterisk/moh folder.

How can I debug this?

Thanks.

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