asterisk users - Aug 2005 - Unable to transfer external calls to MeetMe conference

I have a peculiar situation, and am hoping someone on the list can offer
assistance.  I am running CVS HEAD, and am using ITSPs for DIDs.  The server
has no Zap hardware, but is configured to use ztdummy.  All incoming calls
are via IAX2.  

Calls ring to SIP phones, voice mail, IVR, etc., with no trouble.  I am also
able to transfer calls among my SIP devices, voice mail, IVR, etc.  All of
my SIP devices are able to call into a MeetMe conference without issue.
However, when I attempt to transfer an inbound call from one of my SIP
devices to a MeetMe conference, the call is dropped. If I complete the
transfer while the "You are currently the only person in this
conference"
prompt is playing, the call will successfully make it into the MeetMe
conference, and remains without trouble.  That is the ONLY circumstance in
which I have been able to transfer an external user into the conference.
Also, If I point a DID to the conference in extensions.conf, the call will
ring right into the conference without trouble.  

As an aside, I created a few MOH queues and some corresponding extensions,
so users may hear the music.  When I try to transfer an external call to any
of these MOH extensions, the external caller either hears silence, or the
call is dropped.  Either way, they never hear the MOH.  I do not know if
this is related, but I thought I would mention it.  

I have included CLI output below.  Any assistance will be greatly
appreciated.  

		Sincerely,
		Trevor Hammonds



---- Console output ----

    -- Accepting UNAUTHENTICATED call from x.x.x.x:
       > requested format = ulaw,
       > requested prefs = (ulaw),
       > actual format = ulaw,
       > host prefs = (),
       > priority = caller
    -- Executing Goto("IAX2/xxx@xxx-3", "default|4500|1") in
new stack
    -- Goto (default,4500,1)
    -- Executing SetMusicOnHold("IAX2/xxx@xxx-3",
"ultra-lounge") in new
stack
    -- Executing Set("IAX2/xxx@xxx-3", "Mailbox=4500") in
new stack
    -- Executing Dial("IAX2/xxx@xxx-3", "SIP/4500|20|t") in
new stack
    -- Called 4500
    -- SIP/4500-b9aa is ringing
    -- SIP/4500-b9aa answered IAX2/xxx@xxx-3
    -- Started music on hold, class 'ultra-lounge', on IAX2/xxx@xxx-3
    -- Executing SetMusicOnHold("SIP/4500-98b6",
"ultra-lounge") in new
stack
    -- Executing MeetMe("SIP/4500-98b6", "8600|Ms") in new
stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600'
    -- Playing 'conf-onlyperson' (language 'en')
    -- Started music on hold, class 'ultra-lounge', on SIP/4500-98b6
    -- Stopped music on hold on SIP/4500-98b6
    -- Stopped music on hold on IAX2/xxx@xxx-3
Aug 18 22:14:55 WARNING[24383]: app_meetme.c:841 conf_run: Error getting
conference
    -- Hungup 'Zap/pseudo-2091567275'
  == Spawn extension (from-sip, 8600, 2) exited non-zero on
'IAX2/xxx@xxx-3'
    -- Hungup 'IAX2/xxx@xxx-3'
  == Spawn extension (default, 4500, 3) exited non-zero on
'SIP/4500-98b6<ZOMBIE>'

Trevor G. Hammonds wrote on Thursday, 18 August 2005 10:43 PM:
> I have a peculiar situation, and am hoping someone on the list can
> offer assistance.  I am running CVS HEAD, and am using ITSPs for
> DIDs.  The server has no Zap hardware, but is configured to use
> ztdummy.  All incoming calls are via IAX2.   
> 
> Calls ring to SIP phones, voice mail, IVR, etc., with no trouble.  I
> am also able to transfer calls among my SIP devices, voice mail, IVR,
> etc.  All of my SIP devices are able to call into a MeetMe conference
> without issue.   
> However, when I attempt to transfer an inbound call from one of my
> SIP devices to a MeetMe conference, the call is dropped. If I
> complete the transfer while the "You are currently the only person in
> this conference"   
> prompt is playing, the call will successfully make it into the MeetMe
> conference, and remains without trouble.  That is the ONLY
> circumstance in which I have been able to transfer an external user
> into the conference.   
> Also, If I point a DID to the conference in extensions.conf, the call
> will ring right into the conference without trouble. 
> 
> As an aside, I created a few MOH queues and some corresponding
> extensions, so users may hear the music.  When I try to transfer an
> external call to any of these MOH extensions, the external caller
> either hears silence, or the call is dropped.  Either way, they never
> hear the MOH.  I do not know if this is related, but I thought I
> would mention it.     
An update...  I added an analogue FXO card to this Asterisk server, and
calls coming in on the Zap channels are able to be transferred to the
conference.  However, they do not hear MOH at all (in the conference or
otherwise).  

I have to be missing something.  Hopefully it is obvious to one of you,
because it is obviously escaping me.  ;-)


		Sincerely,
		Trevor Hammonds