[Asterisk-Users] DTMF recognition unreliable or absent, ringing lost on 2nd half of "3-way" call.

[Dave Grey]

These appear to be a common problems, but after spending half a day  
reading the wiki and list archives I have not gained much useful  
knowledge beyond the fact that these are a common problems.  I am  
hoping for some suggestions or pointers to further info.

I have an ivr in my incoming context that does a background() and...  
well, it is an ivr, no need to explain that, I guess.

So, testing locally, it works wonderfully.  Testing through my DID,  
provided by IPKall, it is decidedly hit-or-miss.  The digits seem to  
be either not recognized at all or recognized incorrectly better than  
half the time. Most often, I get the "invalid extension" playback  
that I have assigned to the i,1 exten.  For a while, I had two test  
extensions, one 2000 and one 2001.  Dialing 2001 usually sent me to  
2000 instead. What is making it hard for me to debug is that it  
*sometimes* works, recognizing the extension I dialed correctly.

My peer entry in sip.conf for IPKall contains dtmfmode=rfc2833 as per  
their recommendation.  I have tried setting relaxedtmf=yes in the  
general section, with no noticeable change.  I turned it off again,  
since the problem seems to be too much relaxation in any case.   
Looking at the console, I dial 7056 and it sees 7055, I dial 7056  
again and it sees 75, I dial 7056 a third time and it sees 7055556,  
etc.  Seems random and all over the place.  Packet loss and/or  
ordering?  Aside from the dtmf issue, incoming calls on the DID work  
fine and sound excellent.

Another issue that may or may not be related, but that I would like  
to solve, is that when I flash the switch to initiate a three-way  
call and dial a number, when I flash back to the original call the  
ringing on the second call stops.  I just hear silence until the call  
connects.  When the call does connect, I can send no dtmf at all to  
whatever is at the other end.  To put it another way...  You call  
me.  I want to play you a message on my voicemail, say at the  
office.  I flash the hook and get a dial tone, dial my work VM  
number, and the call starts ringing.  I flash back, and the ringing  
stops.  We listen to the silence together until the VM system picks  
up, but at that point neither of us can send dtmf to log in. (The  
call works normally otherwise, audio in both directions, etc.)

I am sure the answers to these questions require only a basic  
understanding of the way signaling and bridging work over and across  
the different technologies, but I am having a really hard time  
acquiring that understanding.  I would be grateful for any help.

lyd