[asterisk-users] Music on Hold

Wed Sep 16 17:40:11 CDT 2009

This might be another piece of the puzzle:

It would appear any application using playback functionality exits
immediately. For example anything involving voicemail or playback. Phone
calls work with no problem but not if asterisk must play something back.

The modules are loaded however...

Tsunami*CLI> module show like voicemail
Module                         Description                              Use
Count
app_voicemail.so               Comedian Mail (Voicemail System)
0

I'm begining to think that the problem lies with my vendor's package.

On Wed, Sep 16, 2009 at 5:30 PM, Dan Saul <daniel.saul at gmail.com> wrote:

> The files used to be "Frederic Chopin – Polonaised Op. 40-2.raw" I have
> since replaced the raw files with the original mp3s They are now as follows:
>
> [root at Tsunami musiconhold]# ls -l .
> total 13320
> -rw-r--r-- 1 asterisk asterisk 5415926 2009-09-16 17:18 hm1.mp3
> -rw-r--r-- 1 asterisk asterisk 8217974 2009-09-16 17:18 hm2.mp3
>
> I also have the same issue with the default files in /var/lib/asterisk/moh
> .
>
>
> On Wed, Sep 16, 2009 at 5:02 PM, Danny Nicholas <danny at debsinc.com> wrote:
>
>>  What are your actual file names (/etc/asterisk/musiconhold/Frederic
>> Chopin – Polonaised Op. 40-2.wav?)
>>
>>
>>  ------------------------------
>>
>> *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 4:50 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 <danny at 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-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/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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