asterisk users - May 2005 - R: R: AT-320 + supervised transfer

The procedure that will do asterisk is very nice ;) but whe it was available ?

Currently is there any way to emprove the transfer? I tryied the scenario that u
suggest me but it doesn't work :| and i don't why.

Here my sip.conf for the phone, can u say me if there is somethingh wrong ?

[2391]
type=friend
username=2391
secret=2391
language=it
host=dynamic
context=intern
dtmfmode=rfc2833
callgroup=1
pickupgroup=1

I think is ok, maybe i have some problem on phone settings.Can I see your exmple
phone setting ?

Thanks,
Giordano                    
                     

-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill
Inviato: luned? 30 maggio 2005 18.04
A: asterisk-users@lists.digium.com
Oggetto: Re: R: [Asterisk-Users] AT-320 + supervised transfer

On Monday 30 May 2005 16:19, Giordano Grandis wrote:
> Hi,
> Thanks for yuor answer.
>
> The boot time of the phone is very very fast, 10 sec to startup and 2 
> or 3 second to login to asterisk. I set the NTP server to 
> 255.255.255.255 so it don't try to get time.
Well, I run a local NTP server, so it's as fast plus has the correct time at
the end :)
> I thinked carefully to your scenario and i am going to try it, but i 
> don't known if it could like to my customer
>
> I will try also to use CVS, but i am skeptic to utilize asterisk to 
> emprove atxfer...how asterisk emprove the atxfer ?  :|
When Asterisk does the transfer natively, the procedure is like this:

Call comes in, "hold on I'll try to transfer you". you dial *2 (or
any sequence you define), speak to the remote party. If they want to speak to
the caller, YOU hang up. If they don't, THEY hang up and you are returned to
the original caller :)
> How do u set your sip.conf for the at-320 ?  Did u set the
"canreinvite"
> option ?
[1300]
type=friend
username=1300
secret=<ahem>
host=dynamic
context=from-ip
nat=yes
canreinvite=no

Cheers,
Gavin.
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On Monday 30 May 2005 17:22, Giordano Grandis wrote:
> The procedure that will do asterisk is very nice ;) but whe it was
> available ?
Asterisk's atxfer support is only in CVS.
> Currently is there any way to emprove the transfer? I tryied the scenario
> that u suggest me but it doesn't work :| and i don't why.
You *must* be using a new firmware for the phone. Download 1.43 from 

http://www.aredfox.com/edownloadssip.htm

(the AT-320 needs PA186S code)
> Here my sip.conf for the phone, can u say me if there is somethingh wrong ?
Looks fine to me..
> I think is ok, maybe i have some problem on phone settings.Can I see your
> exmple phone setting ?
They're at work so I can't see the config right now... but they're
just the
defaults with the DTMF changed to RFC2833 and the NTP server set... 

Try resetting to defaults using the procedure at 
http://www.voip-info.org/wiki-ATCOM+AT-320

Cheers,
Gavin.

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@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@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@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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill
Inviato: mercoled? 1 giugno 2005 14.20
A: asterisk-users@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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