Asterisk/FreePBX "If you know the extension dial it at any time" issues
In a stock FreePBX/Asterisk/CentOS installation we use a standard main IVR with the phrase " if you know your party's extension dial it at any time". Unfortunately that does not work well. We use three digit extensions all in the 201-250 range. If you dial the digits quickly, you get taken to the invalid entry prompt. If you dial them slowly but with only short gaps between the digits, you are taken to the invalid entry prompt. It seems you must dial one digit, pause, dial the next, pause, and dial the third. Very frustrating!
How can I make this work with speedy fingers and slow fingers alike? I have disabled the IVR choice '2' so as not to confuse, but it is still very touchy. There must be a TIMEOUT(digit) setting somewhere, but since it is FreePBX I am hesitant to start wacking away at the conf files.
I am open to any reasonable solution, but need to sort this out as it is embarrassing!
Thanks!
How can I make this work with speedy fingers and slow fingers alike? I have disabled the IVR choice '2' so as not to confuse, but it is still very touchy. There must be a TIMEOUT(digit) setting somewhere, but since it is FreePBX I am hesitant to start wacking away at the conf files.
I am open to any reasonable solution, but need to sort this out as it is embarrassing!
Thanks!
ASKER
All the calls come in on a VoIP trunk. Will try to find the DTMF and codec settings, thanks.
Your provider is probably maintaining the audio as they receive it which means if you are using one of the higher compression audio codecs as these can distort the audio slightly it can bake dtmf detection more difficult. Switching to g711 (ulaw or alaw) may help in this case.
The best solution would be if the voip providers system listrens into the call as it comes in and provides dtmf detection at that point. SIP is able to signal the DTMF information digitally so that is a more reliable system but does increase the load on the providers system so many may not offer the option or it may be a chargeable extra.
The best solution would be if the voip providers system listrens into the call as it comes in and provides dtmf detection at that point. SIP is able to signal the DTMF information digitally so that is a more reliable system but does increase the load on the providers system so many may not offer the option or it may be a chargeable extra.
ASKER
Sorry, got tied up with some other issues over the weekend. We apparenly allow alaw and ulaw and no others looking at the config files.
Not sure how to get my trunk provider to listen and decode the DTMF, is that a standard feature/request?
Will poke at it some more tomorrow or wednesday, thanks!
Not sure how to get my trunk provider to listen and decode the DTMF, is that a standard feature/request?
Will poke at it some more tomorrow or wednesday, thanks!
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ASKER
I shortened the digit timeout to 3, does not seem to be a problem any more.
ASKER
The comments were very helpful in pursuing the problem, which ultimately led me to the digit timing change which appears to have solved it. So not a complete solution in one (does that really ever happen?), but very helpful and instrumental in resolving it.
If its via an analogue phone line then it can be DTMF tone detection problems.
If it is via VoIP then it can be a combination of codec choice and DTMF tone transmission method.