[Asterisk-Users] Unable to transfer external calls to MeetMeconference (re-post)

[Trevor G. Hammonds]

This message was just bounced back to me.  I am not sure if it made
it to the list originally or not, as I received no responses. 

Since this message was written, I have installed Zap hardware into
this server.  The Zap channels can be transferred to the Meetme
conference.  The IAX2 calls still cannot.  

Any suggestions will be greatly appreciated.

		Sincerely,
		Trevor Hammonds


Trevor G. Hammonds wrote:

> 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 IAX 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 at xxx-3", "default|4500|1") in new
> stack 
>     -- Goto (default,4500,1)
>     -- Executing SetMusicOnHold("IAX2/xxx at xxx-3", "ultra-lounge") in
> new stack 
>     -- Executing Set("IAX2/xxx at xxx-3", "Mailbox=4500") in new stack
>     -- Executing Dial("IAX2/xxx at xxx-3", "SIP/4500|20|t") in new stack
>     -- Called 4500
>     -- SIP/4500-b9aa is ringing
>     -- SIP/4500-b9aa answered IAX2/xxx at xxx-3
>     -- Started music on hold, class 'ultra-lounge', on IAX2/xxx at 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 at 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 at xxx-3' -- Hungup 'IAX2/xxx at xxx-3'
>   == Spawn extension (default, 4500, 3) exited non-zero on
> 'SIP/4500-98b6<ZOMBIE>'