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Giordano Grandis

2005-May-30 10:21 UTC

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

I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for
italian IVR an HTTP pgaes.

So i can only update asterisk with CVS and try atxfer.

Thanks for all

-----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 18.40
A: asterisk-users@lists.digium.com
Oggetto: Re: R: R: [Asterisk-Users] AT-320 + supervised transfer

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.
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Giordano Grandis

2005-May-31 06:41 UTC

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

Hi Gavin,
I installed the cvs Asterisk CVS-D2005.05.28.22.00.00-05/31/05-14:25:23 and i
added this rowd in the features.conf

[featuremap]
blindxfer => #1                ; Blind transfer
disconnect => *0               ; Disconnect
automon => *1                  ; One Touch Record
atxfer => *22                   ; Attended transfer

But...how atxfer work ?

Thanks
                    
                     

-----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 18.40
A: asterisk-users@lists.digium.com
Oggetto: Re: R: R: [Asterisk-Users] AT-320 + supervised transfer

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.
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
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Giordano Grandis

2005-Jun-01 08:44 UTC

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

I did it...but with no good results.
Could i see a example of peer in extensions.conf ? 
I'm trying everythinghs but i always have differenta results :|

Thanks
giordano
                     

-----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 15.31
A: asterisk-users@lists.digium.com
Oggetto: Re: R: R: [Asterisk-Users] AT-320 + supervised transfer

On Wednesday 01 June 2005 14:15, Giordano Grandis wrote:> This is what happen when i call a peer that not answer:
> 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?
I don't like the look of this part at all. Please try to "rm
/usr/lib/asterisk/modules/*" then 'make clean; make install' on a
fresh checkout of CVS HEAD :)

Also, there should be no need for the 'r' option to Dial since SIP
already supports all the progress indication necessary.

gdh
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Giordano Grandis

2005-Jun-06 09:40 UTC

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

Hi Gawin,
I'm newly testing the atxfer and i always the same question: if i transfer a
call to a peer that don't answer me, ho can i re-take the call.
Actually i got the call hanged up without the possibility the speack back with
my first caller.

Thanks
Giordano                    
                     

-----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 15.31
A: asterisk-users@lists.digium.com
Oggetto: Re: R: R: [Asterisk-Users] AT-320 + supervised transfer

On Wednesday 01 June 2005 14:15, Giordano Grandis wrote:> This is what happen when i call a peer that not answer:
> 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?
I don't like the look of this part at all. Please try to "rm
/usr/lib/asterisk/modules/*" then 'make clean; make install' on a
fresh checkout of CVS HEAD :)

Also, there should be no need for the 'r' option to Dial since SIP
already supports all the progress indication necessary.

gdh
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
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Gavin Hamill

2005-Jun-06 12:49 UTC

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

Hi again :)

I'm afraid I simply don't have any more suggestions... It 'works for
me' ...

 gdh

-----Original Message-----
   >From: "Giordano Grandis"<g.grandis@invidea.it>
   >Sent: 06/06/05 17:40:52
   >To: "Asterisk Users Mailing List - Non-Commercial
Discussion"<asterisk-users@lists.digium.com>
   >Subject: R: R: R: [Asterisk-Users] AT-320 + supervised transfer
     >Hi Gawin,
   >I'm newly testing the atxfer and i always the same question: if i
transfer a call to a peer that don't answer me, ho can i re-take the call.
   >Actually i got the call hanged up without the possibility the speack back
with my first caller.
   >
   >Thanks
   >Giordano                    
   >                     
   >

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