asterisk users - Apr 2004 - IAX2 Trunk to PSTN (voicepulse) questions...

All,

I've almost got my Asterisk PBX setup, but I've having some problems
with
the VoicePulse IAX trunk.

On outbound calls, when dialing a PSTN number through the IAX2 trunk,
music on hold (moh, using the m option in the dial command) does not work.
The console states that "stop sound" on IAX2 channel.  Ring works, but
only without the r option.  MOH works when trying to dial a non-PSTN
terminated IAX2 calls (e.g. a softphone).  I've read that with SIP
connetions, the originating line is not held open by the PBX, so the can
be no timing sync with the client, but I don't know if that's also the
case here.

The setup I have is:

[sip softphone Xten] ==> [ * ] ==> [IAX2 VoicePulse Trunk] => [PSTN
Number
(SprintPCS Cell)]

The relevant iax.conf sections are:

[voicepulse]
context=voicepulse-incoming
dtmfmode=rfc2833
secret=mysecret
auth=md5
type=user
host=gw5.voicepulse.com

[voicepulse-peer]
qualify=yes
trunk=yes
dtmfmode=rfc2833
secret=mysecret
auth=md5
type=peer
host=gw5.voicepulse.com

My extensions.conf has:

TRUNK=IAX2/mylogin@voicepulse-peer

exten => 15,1,Playback(transfer)
exten => 15,2,Dial(IAX2/ckm,20,rt)
exten => 15,3,VoiceMail(u${EXTEN})
exten => 15,4,Hangup
exten => 15,103,Dial(${TRUNK}/14155551212,30,t)
exten => 15,104,VoiceMail(u${EXTEN})
exten => 15,105,Hangup

Any ideas, bug?

Thx.

Chris.