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/[EMAIL PROTECTED],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/[EMAIL PROTECTED]/n do you have chan_local?
-- Stopped music on hold on SIP/401-4de6
== Spawn extension (local, 406, 1) exited non-zero on 'Local/[EMAIL
PROTECTED],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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill
Inviato: mercoled� 1 giugno 2005 14.20
A: [email protected]
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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