R: R: [Asterisk-Users] AT-320 + supervised transfer

[Giordano Grandis]

This is what happen when i call a peer that not answer:

   -- Executing Dial("SIP/401-4de6", "SIP/402|60|Thtr") in new stack
    -- Called 402
    -- SIP/402-fa23 is ringing
    -- SIP/402-fa23 answered SIP/401-4de6
    -- Attempting native bridge of SIP/401-4de6 and SIP/402-fa23
    -- Started music on hold, class 'default', on SIP/401-4de6
    -- Playing 'pbx-transfer' (language 'it')
    -- Executing Dial("Local/406 at local-fd88,2", "SIP/406|60|Tthr") in new stack
    -- Called 406
    -- SIP/406-aa46 is ringing
Warning, flexibel rate not heavily tested!
Jun  1 13:45:57 WARNING[25325]: res_features.c:858 builtin_atxfer: Unable to create channel Local/406 at local/n do you have chan_local?
    -- Stopped music on hold on SIP/401-4de6
  == Spawn extension (local, 406, 1) exited non-zero on 'Local/406 at local-fd88,2'
    -- Playing 'beeperr' (language 'it')
  == Spawn extension (local, 402, 1) exited non-zero on 'SIP/401-4de6'

It could some extensions.conf problem ?

Thanks                     

-----Messaggio originale-----
Da: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Per conto di Gavin Hamill
Inviato: mercoledì 1 giugno 2005 14.20
A: asterisk-users at lists.digium.com
Oggetto: Re: R: [Asterisk-Users] AT-320 + supervised transfer

On Wednesday 01 June 2005 13:04, Giordano Grandis wrote:
> Ok, thanks for all.
> Just a thingh: how do u set DTMF on your phones ?

We have them set to RFC2833. 

I think I've noticed some cases where the remote party hears the tones, but it's not an issue that bothers me :)

Cheers,
Gavin.
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