On Wed May 16 2007 23:32:57 Syd Carter wrote: > 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 > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED]
Thanks for the insight. I did pass my concerns with DTMF to my voip provider & I think they fixed up dtmf levels or something; at least dtmf problems seem to be gone. I did not find out what they did to fix it.... mark
