I´m using 2 Digium cards with 4 fxo ports (pluged to gsm gateways) in a FreePBX 2.2.3. I created a DISA so users can dial from cell phones into system throgh zap trunk and postdial using sip trunk. I set a route matching zaptrunk/caller id an it passe in ok the problem is when i dial for example 12083244402 the system understand´s i dialed 00122083332444400022 or 0012084442 or anything else.
Some people told me tha this cant work with gsm gateways because are analogic lines and dont undestand dtmf but i have seen it working and i´m sure it can work if i know how to change the dtmf mode in the trunk and/or disa.
I would thank any help you can give me
It sounds like perhaps the gateway is passing bad audio to the freepbx system (eg, perhaps it’s cutting out during the middle of a keypress, causing * to hear it as two separate presses). First step is probably to route the call to your extension and listen to the DTMF being passed. There is no dtmfmode for zap lines … they’re always “inband” because there is no other way.
Greg MacLellan - Core FreePBX Developer
http//freepbx.org - irc.freenode.net #freepbx
Thanks for the advice I´ll try it and let you know, but what coul be causing this bad audio? and can i fix it somehow?.
Thanks again fo the help
I tested as you told me listening to the dtmf tones through an extension an they dont come in clean at all. it looks like there is a lot of noise and when i press a key in not a continuous tone. May the problem be in the digium cards, in the pc board or just in the gsm lines itself?
Thanks again for your time