Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) <-> Sipura 3k (fxo) <-> Asterisk <-IAX-> remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the "right" way). The attendant answers and asks them the remote extension they want to dial. This setup has worked very well for several months. Last week I upgraded to 1.2.7.1 from 1.2.4 (I think). Since then I've been having trouble with the auto-attendant correctly detecting DTMF (missing digits). Some times it works flawlessly, others I have to try over and over before it is detected correctly. It isn't even consistently dropping the same digit from what I can see on the console. The only thing I've found is that I have a better chance of it working if I wait for the prompt to finish before dialing. I have changed the DTMF method from rfc2833 to info and finally inband with only a little change (inband seems to work the best). Has anyone else run into similar problems or have any more suggestions to try? This is the attendant portion of my extensions.conf: [inter-attendant] exten => s,1,Answer exten => s,2,Wait(1) exten => s,3,Set(TIMEOUT(response)=10) exten => s,4,Background(enter-ext-of-person) exten => i,1,Playback(invalid) exten => i,2,Goto(s,4) exten => i,3,Hangup exten => t,1,Playback(goodbye) exten => t,2,Hangup include => tests include => fullertonpbx include => intercompany Thank you for any insight you can provide. Dave Fullerton
This is a known issue with Asterisk's implementation of DTMF detection. There are two bug reports open up on bug tracker. Currently the best combination is to set DTMF TX method on Spa3k to INFO and auto on asterisk side. Works 95%, skips digits if you press buttons on the FXO end too fast. Until the DTMF stuff gets rewritten in Asterisk this gonna be this way, so far 2 years still no 100% dtmf detection, both detection and transmit parts are flawed and dont work 100%, even inband. Affected fxo gateways are (tested by myself): Sipura, Addpac, Planet, Wellgate. HTH, Vahan Dave Fullerton wrote:> > Greetings, > > I'm using asterisk to connect our three locations together with a sort > of inter-company auto attendant connected like this: > > PBX (fxs) <-> Sipura 3k (fxo) <-> Asterisk <-IAX-> remote asterisk > > It works like this: Person picks up their phone and dials a number to > get to the auto attendant (I don't have any FXO ports available on our > PBX to do it the "right" way). The attendant answers and asks them the > remote extension they want to dial. This setup has worked very well for > several months. Last week I upgraded to 1.2.7.1 from 1.2.4 (I think). > Since then I've been having trouble with the auto-attendant correctly > detecting DTMF (missing digits). Some times it works flawlessly, others > I have to try over and over before it is detected correctly. It isn't > even consistently dropping the same digit from what I can see on the > console. The only thing I've found is that I have a better chance of it > working if I wait for the prompt to finish before dialing. I have > changed the DTMF method from rfc2833 to info and finally inband with > only a little change (inband seems to work the best). > > Has anyone else run into similar problems or have any more suggestions > to try? > > This is the attendant portion of my extensions.conf: > > [inter-attendant] > exten => s,1,Answer > exten => s,2,Wait(1) > exten => s,3,Set(TIMEOUT(response)=10) > exten => s,4,Background(enter-ext-of-person) > > exten => i,1,Playback(invalid) > exten => i,2,Goto(s,4) > exten => i,3,Hangup > > exten => t,1,Playback(goodbye) > exten => t,2,Hangup > > include => tests > include => fullertonpbx > include => intercompany > > > > Thank you for any insight you can provide. > > Dave Fullerton > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
I too am experiencing DTMF problems with 1.2.7.1 that I did not experience with recent prior versions. I've backed up to version 1.2.6 and so far DTMF detection is working reliably (but that's only with about 10 calls worth of testing). I've only had problems over SIP channels. Zap channels did not have problems with 1.2.7.1. I do not have any IAX channels, so cannot comment on that. I know others tend to discount DTMF problems because of "known problems" with how Asterisk handles DTMF, but there does seem to be enough anecdotal evidence that something bad has recently happened to make things worse. Dave, would you mind trying version 1.2.6 to see if that also resolves your problems? Dave Fullerton wrote:> > Greetings, > > I'm using asterisk to connect our three locations together with a sort > of inter-company auto attendant connected like this: > > PBX (fxs) <-> Sipura 3k (fxo) <-> Asterisk <-IAX-> remote asterisk > > It works like this: Person picks up their phone and dials a number to > get to the auto attendant (I don't have any FXO ports available on our > PBX to do it the "right" way). The attendant answers and asks them the > remote extension they want to dial. This setup has worked very well > for several months. Last week I upgraded to 1.2.7.1 from 1.2.4 (I > think). Since then I've been having trouble with the auto-attendant > correctly detecting DTMF (missing digits). Some times it works > flawlessly, others I have to try over and over before it is detected > correctly. It isn't even consistently dropping the same digit from > what I can see on the console. The only thing I've found is that I > have a better chance of it working if I wait for the prompt to finish > before dialing. I have changed the DTMF method from rfc2833 to info > and finally inband with only a little change (inband seems to work the > best). > > Has anyone else run into similar problems or have any more suggestions > to try? > > This is the attendant portion of my extensions.conf: > > [inter-attendant] > exten => s,1,Answer > exten => s,2,Wait(1) > exten => s,3,Set(TIMEOUT(response)=10) > exten => s,4,Background(enter-ext-of-person) > > exten => i,1,Playback(invalid) > exten => i,2,Goto(s,4) > exten => i,3,Hangup > > exten => t,1,Playback(goodbye) > exten => t,2,Hangup > > include => tests > include => fullertonpbx > include => intercompany > > > > Thank you for any insight you can provide. > > Dave Fullerton > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users