[asterisk-users] DTMF Problems

[Brent Vrieze]

First of all Merry Christmas.

Second, my first problem with my provider not staying registered with 
our server was my fault.  We moved our server room and I restarted the 
test system and the production system causing them to ping-pong back and 
forth registering with our provider causing random problems, they are 
both set to register with the same account right now.  I shut Asterisk 
down on the one and now we don't drop any longer.  doh!!!

Last, We are having DTMF problems with our provider (via:talk).  Does 
anyone have any experience with them and if so can you share it?  
via:talk does have a sample sip.conf and extensions.conf file to use but 
the dial plan they set up does not require any DTMF so they may never 
have tested it.  We have tried inband, auto, rfc2833 for our DTMF and 
nothing works.  I have submitted a ticket with them but the last time I 
did that they never responded so that is why I am posting here.
I signed up with another SIP provider for a test account and the DTMF 
passes no problem from them so I must conclude there is some setting 
that via:talk has that is causing the problem.  via:talk will not 
confirm this but they must be using Asterisk as all the menus and such 
they have feel very Asteriskish.  Is there something I can tell via:talk 
to try on their end to make this work?

As a side symptem every time our system registers with via:talk it seams 
to jump from server to server on their end.  They must have some sort of 
load balancing going on that is causing that.  In the past we could get 
the DTMF to pass when we were on the initial server we registered with 
but when we got pushed to another server the DTMF would fail till I did 
a sip reload or restarted Astersk.  Now we get no DTMF ever.

System set up.
Asterisk 1.4.22
Asterisk GUI 2.0

users.conf
[trunk_1]
context = DID_trunk_1
host = galvatron.vtnoc.net
username = user name
secret = password
trunkname = via:talk - galvatron  ; GUI metadata
hasiax = no
registeriax = no
hassip = yes
registersip = yes
trunkstyle = voip
hasexten = no
fromuser = user name
authuser = user name
insecure = port,invite
dtmf = rfc2833
dtmfmode = rfc2833
relaxdtmf = yes
rfc2833compensate = yes
port = 5060
canreinvite = no
fromdomain = galvatron.vtnoc.net
disallow = all
allow = ulaw,gsm

If you need to see more of the setup info I can provide.

Thanks
   Brent