[Asterisk-Users] Iax unable to transfer using Telappliant

[Roy Eddleston]

Update

I have found that setting notransfer=yes enables me to call Telappliant
numbers (09XX) and not get disconnected but if I call a BT number the
call goes out via Telappliant to the BT phone, it rings, the client
answers, they can hear you, but firefly does not know the other end has
been answered and continues to ring, obviously then you cant hear the
client. Is this a symptom of notransfer=yes or is there another problem?
Firewall related maybe?

Anyone with a working Telappliant account using IAX?

Cheers!

Roy.... 

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of Roy Eddleston
> Sent: 28 July 2004 09:13
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Iax unable to transfer
> 
> Dimitri
> 
> Did you get a resolution to this problem? I am seeing the same, my *
box
> talks to Telappliant using AIX, anybody else seen this?
> 
> Roy..
> 
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> > admin at lists.digium.com] On Behalf Of reseaux
> > Sent: 23 June 2004 10:47
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] Iax unable to transfer
> >
> > Dear List
> > 	I have notice this kind of problem between my two * box.
> > My scenario is :
> > 				    Iax GSM
> > IaxClient----->PBX1------------>PBX2-->TDM
> > 			today CVS		Stable V1
> >
> > I use as Client FireFly with IAX2/GSM and try to call my PBX1 this
> server call
> > PBX2 to terminate the call trought a TDM line (TE410P) but after
PBX2
> join
> > the two call i can see the log below from my PBX1, i can speak for
few
> second
> > and after the FireFly hangup.
> > I have try to change * version from Stable to today CVS but no
success
> same
> > problem.
> > I have enabled the IAX Debug and seems the RX side (PBX1) dont
accept
> > something from PBX2 and show the "unable to transfer" (im not
expert)
> :-)
> >
> > The strange thing is if i call from Sip Phone/client i dont have a
> problem the
> > Call is bridged!
> >
> > The events from the CLI:
> > ---------
> > Executing Dial("IAX2[pippo2 at pippo2]/5",
> > "IAX2/out:###@213.215.xxx.xxx/0012234456666 at outgoing|60|g") in new
> stack
> >     -- Called out:######@213.215.xx.xx/0012234456666 at outgoing
> >     -- Call accepted by 213.215.xx.xx (format GSM)
> >     -- Format for call is GSM
> >     -- IAX2[out]/6 stopped sounds
> >     -- IAX2[out]/6 is ringing
> >     -- IAX2[out]/6 stopped sounds
> >     -- IAX2[out]/6 answered IAX2[pippo2 at pippo2]/5
> >     -- Attempting native bridge of IAX2[pippo2 at pippo2]/5 and
> IAX2[out]/6
> >     -- Channel 'IAX2[out]/6' unable to transfer
> >     -- Hungup 'IAX2[out]/6'
> >     -- Executing Hangup("IAX2[pippo2 at pippo2]/5", "") in new stack
> >   == Spawn extension (incoming,0012234456666, 4) exited non-zero on
> 'IAX2
> > [pippo2 at pippo2]/5'
> >     -- Executing Hangup("IAX2[pippo2 at pippo2]/5", "") in new stack
> > -----------
> >
> > Thanks in advance for possible help!
> > Dimitri
> >
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> 
> 
> 
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