For using the non-variable length DTMF in 1.4, you can use:
dtmfmode=rfc2833compensate This is required when utilizing DTMF between Asterisk 1.4 and older versions. This may help resolve some issues you may be having with certain providers. Leif. On 5/16/07, Syd Carter <[EMAIL PROTECTED]> 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]
-- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk
