Hi Mark. Re #1 - I too have be wrestling with DTMF issues ever since
upgrading to 1.4 1.4 specifically made a change to DTMF signalling. I
also have a working 1.2 install and when I revert back to that, the DTMF
issues disappear. Unfortunately I'm too much in love with 1.4 to use
1.2 so I'm trying this and that to determine how to cope. Worse comes
to worse, you could try extending the DTMF duration in the source file
and recompile.
Mark Borg wrote:
Hi, I am a "new member" trying to get up to speed with asterisk.
Got 2 questions:
1. I have a setup at my shop, voip via voicenet.ca; asterisk 1.4 & aastra 480i
terminal. No analog lines avail.
It seems fine, except, for with some calls, I cannot have a dtmf tone
recognised by the called pbx. (like press 0 for operator, etc.) On other
pbx's it works OK.
2. I set up a client with two TDM400 & 6 POTS lines (only)& AAstra 9131
terminals. One application is to bridge an inbound call to another line &
dial back out to a remote office. HOWEVER, the resulting call is at a very
low volume & client is irritated. Changing the line parameters with fxotune
just makes a bunch of clipping & echo, and one cannot tell which lines are to
be bridged as it depends on the incoming call.
Does anyone have insight?
Should I (eat the TDM400s and) install sipura hardware?
thanks
Mark Borg
