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Giordano Grandis

2005-May-30 08:19 UTC

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

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.

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 ?  :|

How do u set your sip.conf for the at-320 ?  Did u set the
"canreinvite" option ?

Thanks for all,
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 16.22
A: asterisk-users@lists.digium.com
Oggetto: RE: [Asterisk-Users] AT-320 + supervised transfer

Hi,

1.0.7 does not support atxfer. You nees to use CVS. I have a bunch of the ATCOM
cheapos and only CVS will take notice of 'atxfer' in features.conf.

Otherwise , consider this scenario...

Call comes in, press HOLD, dial other party to see if they wish to speak to the
caller. If so, press * to hang up, then HOLD to swap back to the incoming
caller. Announce you are going to transfer them, and now dial the final
extension and press FWD to do a blind transfer.

This works for me with the SIP 1.43 firmware. The IAX fw still has some way to
go...

The phone seems slow to boot? Ensure you have the IP/hostname of a valid NTP
time server at the bottom of the web config page. If you don't, it will take
ages and eventually fall back on 'time.windows.com'

Cheers,
Gavin



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Gavin Hamill

2005-May-30 09:03 UTC

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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.

Giordano Grandis

2005-Jun-01 05:04 UTC

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

Ok, thanks for all.
Just a thingh: how do u set DTMF on your phones ?

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 13.51
A: asterisk-users@lists.digium.com
Oggetto: Re: [Asterisk-Users] AT-320 + supervised transfer

On Wednesday 01 June 2005 12:43, Giordano Grandis wrote:> No...maybe i don't explain u well.
>
> After that B call C andC not answer (go in timeout), B hear first the 
> beeperr and then, together A the busy tone. Now i can't re-take the 
> call :|
I'm afraid I don't have any more suggestions to offer - anyone else?

Cheers,
Gavin.
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Kanuri, Seshu (Company IT)

2005-Jun-02 08:40 UTC

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

Remove the Tthr options. You don't need any of them in the dial string for
AT320s

Seshu

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Giordano Grandis
Sent: Wednesday, June 01, 2005 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: R: R: [Asterisk-Users] AT-320 + supervised transfer

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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