Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: Teviot on March 19, 2009, 02:02:26 AM
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I want to test it again to make sure, but I tried to call a government department yesterday that hs one of those horrid menu system ... press 1 for this ... press 2 for that ... press 3 for something else etc.
Like I said I want to test i again but I thought you should know that there might be a problem.
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you can get problems if you have a mismatch between what the phone is sending and what asterisk (and the upstream) is listening for. This is because there are three different ways in which you can send and receive dtmf in the SIP world. If in doubt, set everything to sipinfo (dtmfmode=info) and try again.
Best
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I believe it was working prior to v719. Have added the "dtmfmode=info" into the extensions configs and will see how that goes