asterisk users - Sep 2003 - MusicOnHold (MOH) silent on BudgeTone-100 only.

I have the MusicOnHold feature working great when called from ATA-186
extensions.  It's pretty cool.

However, when I call from a BudgeTone-100 phone, no music is heard --
instead it continues the ringing feedback and acts like the call is
unanswered.  At the same time, I can call from (multiple) ATA-186
extensions and hear music as long as I like.  How can I debug this?
As far as I can tell, Asterisk thinks it's connecting the BudgeTone
channel just fine.  Only it's silent.

Normal voice calls between BudgeTone and ATA-186 extensions work fine,
of course.  They use ulaw encoding.  It's just the MOH that fails to
answer.  There is no NAT or firewall involved -- all hosts are local.

I'm testing with dedicated extensions like this: 

exten => 302,1,WaitMusicOnHold(5)
exten => 303,1,MusicOnHold(loud)
exten => 304,1,MusicOnHold(default)

Here's the asterisk output from a working ATA-186 call:
*CLI>
    -- Executing MusicOnHold("SIP/200-990d", "loud") in new
stack
    -- Started music on hold, class 'loud', on SIP/200-990d

[ beautiful music emanates from earpiece ]

    -- Stopped music on hold on SIP/200-990d
  == Spawn extension (dialout, 303, 1) exited non-zero on 'SIP/200-990d'


And from the silent BudgeTone-100:
    -- Executing MusicOnHold("SIP/202-351f", "loud") in new
stack
    -- Started music on hold, class 'loud', on SIP/202-351f

[ Budgetone still provides ringing indication until I end.]

    -- Stopped music on hold on SIP/202-351f
  == Spawn extension (dialout, 303, 1) exited non-zero on 'SIP/202-351f'


While the BudgeTone is 'ringing', Asterisk appears to think it has a
live connection:

*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter  Format
192.168.42.56    202         26e8973e-e3  00101/06079  00000ms  0000ms  ULAW  
1 active SIP channel(s)

Has anyone else tried MOH from a Grandstream BudgeTone extension?  It
would mean a lot just to hear that it's my local config and not some
weird BT problem.  If someone could post a working BudgeTone SIP
definition, that might help too.

Thanks!
-Steve

It works fine for me, I created a 2nd "music on hold", tossed a bunch
of
mp3 files into a directory and I can listen to music on the
speakerphone:

;radio @ 8888
exten => 8888,1,Answer
exten => 8888,2,MusicOnHold(default)

> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-
> admin@lists.digium.com] On Behalf Of Steve Haehnichen
> Sent: Sunday, September 14, 2003 1:14 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] MusicOnHold (MOH) silent on BudgeTone-100
only.
> 
> pamAssassin 2.55 (1.174.2.19-2003-05-19-exp)
> 
> I have the MusicOnHold feature working great when called from ATA-186
> extensions.  It's pretty cool.
> 
> However, when I call from a BudgeTone-100 phone, no music is heard --
> instead it continues the ringing feedback and acts like the call is
> unanswered.  At the same time, I can call from (multiple) ATA-186
> extensions and hear music as long as I like.  How can I debug this?
> As far as I can tell, Asterisk thinks it's connecting the BudgeTone
> channel just fine.  Only it's silent.
> 
> Normal voice calls between BudgeTone and ATA-186 extensions work fine,
> of course.  They use ulaw encoding.  It's just the MOH that fails to
> answer.  There is no NAT or firewall involved -- all hosts are local.
> 
> I'm testing with dedicated extensions like this:
> 
> exten => 302,1,WaitMusicOnHold(5)
> exten => 303,1,MusicOnHold(loud)
> exten => 304,1,MusicOnHold(default)
> 
> Here's the asterisk output from a working ATA-186 call:
> *CLI>
>     -- Executing MusicOnHold("SIP/200-990d", "loud") in
new stack
>     -- Started music on hold, class 'loud', on SIP/200-990d
> 
> [ beautiful music emanates from earpiece ]
> 
>     -- Stopped music on hold on SIP/200-990d
>   == Spawn extension (dialout, 303, 1) exited non-zero on
'SIP/200-990d'
> 
> 
> And from the silent BudgeTone-100:
>     -- Executing MusicOnHold("SIP/202-351f", "loud") in
new stack
>     -- Started music on hold, class 'loud', on SIP/202-351f
> 
> [ Budgetone still provides ringing indication until I end.]
> 
>     -- Stopped music on hold on SIP/202-351f
>   == Spawn extension (dialout, 303, 1) exited non-zero on
'SIP/202-351f'
> 
> 
> While the BudgeTone is 'ringing', Asterisk appears to think it has
a
> live connection:
> 
> *CLI> sip show channels
> Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter
> Format
> 192.168.42.56    202         26e8973e-e3  00101/06079  00000ms  0000ms
> ULAW
> 1 active SIP channel(s)
> 
> Has anyone else tried MOH from a Grandstream BudgeTone extension?  It
> would mean a lot just to hear that it's my local config and not some
> weird BT problem.  If someone could post a working BudgeTone SIP
> definition, that might help too.
> 
> Thanks!
> -Steve
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

Steve,

I'm certainly not an expert on any of this, but I just spent the better
part of two weeks trying to understand/fix a music on hold problem (my
case was no moh at all). The things that I discovered include:

1. music on hold does not require any "exten => 302,1,..." commands
at
all. Once moh is defined in the musiconhold.conf file, it seems to work
on all extentions without additional per-extension definitions. (I didn't
attempt to research the "exten => ..." approach, but I'd guess
that
statement is used for specific requirements such as playing a certain
mp3 for on a per-extension basis.)

2. when a conversation between two extensions is in progress and one
extension places the call on hold, that extension informs *, and *
establishes a new session with the other extension over which the music
is played.  If you run "sip debug" from the CLI, you will see the
handshaking that creates this new temp session.

Given those two items, it would appear as though your problem is 
related to how the phone itself is configured and not *. I'd look for
a phone configuration command with a keyword something like "redirect"
and use the "sip debug" to see any error messages.

Rich
------------------------
> I have the MusicOnHold feature working great when called from ATA-186
> extensions.  It's pretty cool.
> 
> However, when I call from a BudgeTone-100 phone, no music is heard --
> instead it continues the ringing feedback and acts like the call is
> unanswered.  At the same time, I can call from (multiple) ATA-186
> extensions and hear music as long as I like.  How can I debug this?
> As far as I can tell, Asterisk thinks it's connecting the BudgeTone
> channel just fine.  Only it's silent.
> 
> Normal voice calls between BudgeTone and ATA-186 extensions work fine,
> of course.  They use ulaw encoding.  It's just the MOH that fails to
> answer.  There is no NAT or firewall involved -- all hosts are local.
> 
> I'm testing with dedicated extensions like this: 
> 
> exten => 302,1,WaitMusicOnHold(5)
> exten => 303,1,MusicOnHold(loud)
> exten => 304,1,MusicOnHold(default)
> 
> Here's the asterisk output from a working ATA-186 call:
> *CLI>
>     -- Executing MusicOnHold("SIP/200-990d", "loud") in
new stack
>     -- Started music on hold, class 'loud', on SIP/200-990d
> 
> [ beautiful music emanates from earpiece ]
> 
>     -- Stopped music on hold on SIP/200-990d
>   == Spawn extension (dialout, 303, 1) exited non-zero on
'SIP/200-990d'
> 
> 
> And from the silent BudgeTone-100:
>     -- Executing MusicOnHold("SIP/202-351f", "loud") in
new stack
>     -- Started music on hold, class 'loud', on SIP/202-351f
> 
> [ Budgetone still provides ringing indication until I end.]
> 
>     -- Stopped music on hold on SIP/202-351f
>   == Spawn extension (dialout, 303, 1) exited non-zero on
'SIP/202-351f'
> 
> 
> While the BudgeTone is 'ringing', Asterisk appears to think it has
a
> live connection:
> 
> *CLI> sip show channels
> Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter 
Format
> 192.168.42.56    202         26e8973e-e3  00101/06079  00000ms  0000ms 
ULAW
> 1 active SIP channel(s)
> 
> Has anyone else tried MOH from a Grandstream BudgeTone extension?  It
> would mean a lot just to hear that it's my local config and not some
> weird BT problem.  If someone could post a working BudgeTone SIP
> definition, that might help too.