asterisk users - Jul 2003 - Call transfer on ATA186

Hi,

The call transfer function in Asterisk seems to work in a way which does not
permit to ATA186 (or any othet hardware phone with only pne line)  to have
this feature.
If someone tries to transfer a call to an ATA186 based extension, the call
is transferred to the correspondent voice mailbox, because the first
extension tries to call the last one after the transfer is initiated and
during this time ATA is bussy.

It is something that can be done to solve this issue?
Someone else using Call Transfer on ATA186?
They are some special switches available for Dial command in order to get
Call Transfer workig on ATA?

Thanks,
Dan

The basic call transfer functions, set with the T and t options to the dial 
application and triggered by pressing a # work fine for me.  Make sure that 
you have set the DialPlan on the ATA 186 so as not to grab the # (ie look 
for any ># character pairs and change the second character or remove it).

  Iain





--On Monday, July 28, 2003 6:58 pm +0300 Dan <dtoma@fx.ro> wrote:
> Hi,
>
> The call transfer function in Asterisk seems to work in a way which does
> not permit to ATA186 (or any othet hardware phone with only pne line)  to
> have this feature.
> If someone tries to transfer a call to an ATA186 based extension, the call
> is transferred to the correspondent voice mailbox, because the first
> extension tries to call the last one after the transfer is initiated and
> during this time ATA is bussy.
>
> It is something that can be done to solve this issue?
> Someone else using Call Transfer on ATA186?
> They are some special switches available for Dial command in order to get
> Call Transfer workig on ATA?
>
> Thanks,
> Dan
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

> Hi all ATA-Users,
Hi,
> after a lot of tests, i found the best (not complete working solution).
>
> If you use an an MGCP-Image then
I use SIP image, v2.16.
> 1. CLIP-CallerID works fine (with one Phone Callername-transmission
> works too!!!!)
For me too, including extended caller id info (display on the phone both the
name and the number of the caller).
Default values for Caller ID are used in the configuration page
> 2. Blind transfer with # works fine
For me too, including with another ATA destination. I have tried both semi
and full unattended modes and it works.
> 3. Attended transfer (Transfer with consultation?) works with incoming
> and outgoing calls (with Flash).
For me it works only if the final destination is not an ATA device too. It
is something normal with the actual call transfer implementation in
Asterisk. See my previous mails. If the redial will be done in more than one
second, this will work on ATA as a final destination too. Mark, can this be
implemented in the call trensfer function inside Asterisk?
> I never tried H323 because of the mammut sources to compile.
:-) I have tried H.323, but only with NetMeeting, not with ATA. There is any
reason to use H.323 instead of SIP? They are some features which are
available in H.323 only?
> With SIP there was no way to get all things working.
Can you detail more? What exactly does not work as expected?
> Sometimes the sound (over ATA) is choppy.
I have used ATA(SIP) with a very cheap (around $5) analog phone and the
sound is very good, even better that the one of a Cisco 7960 (this is what
the other party told me). Not even a single interruption. The DTMF
functionality is far better than the one on 7960. Not even a single
miss-detection with ATA.
> Then i have to reboot the ATA and everything is fine
> again (any hints?).
Never have this problem with ATA.
Which firmware version do you use?

Best regards,
Dan

----- Original Message ----- 
From: "Thomas Dingermann" <td@trobisch.de>
To: <asterisk-users@lists.digium.com>
Sent: Tuesday, July 29, 2003 9:38 AM
Subject: Re: [Asterisk-Users] Call transfer on ATA186

> Hi all ATA-Users,
>
> after a lot of tests, i found the best (not complete working solution).
>
> If you use an an MGCP-Image then
>
> 1. CLIP-CallerID works fine (with one Phone Callername-transmission
> works too!!!!)
> 2. Blind transfer with # works fine
> 3. Attended transfer (Transfer with consultation?) works with incoming
> and outgoing calls (with Flash).
>
> I never tried H323 because of the mammut sources to compile. With SIP
> there was no way to get all things working. Sometimes the sound (over
> ATA) is choppy. Then i have to reboot the ATA and everything is fine
> again (any hints?).
>
>
>
> - Thomas
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>