On Feb 27, 2006, at 6:09 AM, Dr. Michael J. Chudobiak wrote:
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always
works fine for an unlimitel.ca account.
Someone else has seen this too: http://bugs.digium.com/view.php?id=6011
Can anyone suggest a workaround (other than jitterbuffer=off)?
Actually I don't think Asterisk should jitter buffer in the above case?
I am a newb, so be warned.
My research seems to indicate that jitter buffering should only be used
at the end points, as that is where the audio needs to be reassembled.
Since in this case asterisk is the man in the middle and not one of the
endpoints (I think?) it doesn't need to jitter buffer at all for calls
being placed through an outside IAX carrier? If what I have written is
true, then jitter buffering is only adding extra latency.
If you are using Zap channels the above is probably wrong though?
I noticed also, from one of my handsets attached to an ATA (AG168v)
connecting through IAX2, DTMF was sensitive to volume adjustment even
though it is out of band (rfc2833).
Another thing that might help in the case you describe is to use a more
band width efficient codec like G729 or GSM versus uLaw or alaw.
Sorry if this is all old hat to you and I am restating the obvious.
Marty
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