Re: Asterisk Hardware - ATA

2007-10-24 Thread ik
On 10/24/07, sammy ominsky <[EMAIL PROTECTED]> wrote:
> Lots of good info to be found at http://www.voip-info.org/

And you can also download the book "Astersiks the feature of
telephony" 2nd edition and read it.
http://www.cyberciti.biz/tips/download-asterisk-howto-tutorial-book.html

It contain mostly up to date information some of it was taken from
voip-info.org.

>
> ---sambo
>
>
> On 24/10/2007, at 08:21, Eran Levy wrote:
>
> > Is there any good guide/tutorial for Asterisk configuration &
> > management?
> > I'm facing many issues/requests and I can't handle it in a short
> > time frame...
> >
> > Eran
> >
> > On 10/24/07, Meir Michanie <[EMAIL PROTECTED]> wrote: I see that
> > posting about asterisk interested many people in the list as
> > I see nobody complained so far.
> > I would like to give my tip from my experience with grandstream ata
> > 488.
> > I bought it from Tikal networks. It came with  a very  old firmware
> > that
> > caused  me a lot of trouble.
> > 1. It would disconnect phonecalls ramdomly.
> > 2. there was no bridge mode between interfaces.
> > 3. very low quality and echo.
> > 4. the caller ID from the PSTN is not forwarded to the sip channel.
> > 5. calling through the FXO did not work as a trunk but I would have to
> > call dial (SIPEXT,timeout,D(pstndest).
> > 6. forced to ring once on the local phone before forwarding call to
> > sip.
> >
> > Today I upgraded to the latest firmware and now all issues beside the
> > caller id were solved. Shame that it didn't come with the latest
> > firmware, I would have a better taste on my mouth but may be lower
> > learning curve.
> >
> > One more post.
> >
> >
> >
> >
> > --
> > Thanks,
> > Eran
>


Ido
-- 
http://ik.homelinux.org/

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Re: Asterisk Hardware - ATA

2007-10-23 Thread sammy ominsky

Lots of good info to be found at http://www.voip-info.org/

---sambo


On 24/10/2007, at 08:21, Eran Levy wrote:

Is there any good guide/tutorial for Asterisk configuration &  
management?
I'm facing many issues/requests and I can't handle it in a short  
time frame...


Eran

On 10/24/07, Meir Michanie <[EMAIL PROTECTED]> wrote: I see that  
posting about asterisk interested many people in the list as

I see nobody complained so far.
I would like to give my tip from my experience with grandstream ata  
488.
I bought it from Tikal networks. It came with  a very  old firmware  
that

caused  me a lot of trouble.
1. It would disconnect phonecalls ramdomly.
2. there was no bridge mode between interfaces.
3. very low quality and echo.
4. the caller ID from the PSTN is not forwarded to the sip channel.
5. calling through the FXO did not work as a trunk but I would have to
call dial (SIPEXT,timeout,D(pstndest).
6. forced to ring once on the local phone before forwarding call to  
sip.


Today I upgraded to the latest firmware and now all issues beside the
caller id were solved. Shame that it didn't come with the latest
firmware, I would have a better taste on my mouth but may be lower
learning curve.

One more post.


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--
Thanks,
Eran



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Re: Asterisk Hardware - ATA

2007-10-23 Thread Eran Levy
Is there any good guide/tutorial for Asterisk configuration & management?
I'm facing many issues/requests and I can't handle it in a short time
frame...

Eran

On 10/24/07, Meir Michanie <[EMAIL PROTECTED]> wrote:
>
> I see that posting about asterisk interested many people in the list as
> I see nobody complained so far.
> I would like to give my tip from my experience with grandstream ata 488.
> I bought it from Tikal networks. It came with  a very  old firmware that
> caused  me a lot of trouble.
> 1. It would disconnect phonecalls ramdomly.
> 2. there was no bridge mode between interfaces.
> 3. very low quality and echo.
> 4. the caller ID from the PSTN is not forwarded to the sip channel.
> 5. calling through the FXO did not work as a trunk but I would have to
> call dial (SIPEXT,timeout,D(pstndest).
> 6. forced to ring once on the local phone before forwarding call to sip.
>
> Today I upgraded to the latest firmware and now all issues beside the
> caller id were solved. Shame that it didn't come with the latest
> firmware, I would have a better taste on my mouth but may be lower
> learning curve.
>
> One more post.
>
>
> =
> To unsubscribe, send mail to [EMAIL PROTECTED] with
> the word "unsubscribe" in the message body, e.g., run the command
> echo unsubscribe | mail [EMAIL PROTECTED]
>
>


-- 
Thanks,
Eran


Asterisk Hardware - ATA

2007-10-23 Thread Meir Michanie
I see that posting about asterisk interested many people in the list as
I see nobody complained so far.
I would like to give my tip from my experience with grandstream ata 488.
I bought it from Tikal networks. It came with  a very  old firmware that
caused  me a lot of trouble.
1. It would disconnect phonecalls ramdomly.
2. there was no bridge mode between interfaces.
3. very low quality and echo.
4. the caller ID from the PSTN is not forwarded to the sip channel.
5. calling through the FXO did not work as a trunk but I would have to
call dial (SIPEXT,timeout,D(pstndest).
6. forced to ring once on the local phone before forwarding call to sip.

Today I upgraded to the latest firmware and now all issues beside the
caller id were solved. Shame that it didn't come with the latest
firmware, I would have a better taste on my mouth but may be lower
learning curve.

One more post.


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