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
