Hi All,
I m using asterisk 1.2.24 on CentOS HP Proliant ML-150, i m facing problem with dtmf tones which are passing through. I dial a IVR after connecting i press the keys but ivr goes on no more key acceptance. Can any one tell how can i resolve the issue.
regards…
Atif shahzad
Hi
Sounds like you have the DTMF mode wrong. check with your trunk supplier and check the peer settings as well
Ian
my sip.conf settings are:
[test]
type=peer
host=X.X.X.X
username=test123
secret=2001
fromuser=test123
canreinvite=no
disallow=all
allow=g729
allow=ulaw
allow=g723
dtmfmode=rfc2833
and according to them these settings are OK.
Atif
give it a try?
With these settings m getting problem i change dtmf to inband, auto with no luck, can you help me out this issue…
Atif
Hi
What handset are you using ? and what codec is established when the call is up
Ian
m using Eyebeam, n when the call is established it shows g729.
Atif
Do you have G729 codecs ? try changing the sip.conf to
[test]
type=peer
host=X.X.X.X
username=test123
secret=2001
fromuser=test123
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
Also make sure the DTMF mode of the eyebeam peer match as well
but i will consume much bandwidth with 30 agents i have only 1MB E1 link. What you suggest?
Atif
Hi I Know it will, but this is for testing, are the softphones remote to the server ?
also what does the sip.conf entry for the sofphones looklike
Ian
[2002]
disallow=all
allow=g729
allow=ulaw
allow=alaw
host=dynamic
type=friend
username=2002
secret=test
dtmfmode=rfc2833
canreinvite=no
context=default
Here is my user settings for Eyebeam user in sip.conf
Atif