[Asterisk-Users] Call transfer on ATA186

[Dan]

> 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 at trobisch.de>
To: <asterisk-users at 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 at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>