Pierre,
Thats exactly what Joanna said in her reply.
Check the client DTMF settings on your phones.
set it to rfc2833 or out-of-band, whatever the config says.
Grandstream by default have inband DTMF set, and usualy ulaw is
supported as well, and thats the reason ur grandstream works but others
dont.
cheerz
- Ben.
Pierre Marceau wrote:
Hi Joanna,
Thanks for your reply.
In my mind I think it must be some setting in the client (phone) becasue the
Grandstream GXP 2000 does work and it is using the same sip.conf
Extensions:
6000 is xlite softfone
6003 is Linksys SPA941
6004 is Grandstream GXP 2000
6005 is Linksys PAP2NA
Please have a look at my sip conf and suggest any changes I could try...
[general]
context=internal
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
type=friend
secret=XXXXXXX
nat=no
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
subscribecontext=internal
canreinvite=no
register=8885551234:[EMAIL PROTECTED]
[atlasvoice]
type=friend
host=proxy.atlasvoice.com
username=8885551234
secret=XXXXXXX
fromuser=8885551234
fromdomain=proxy.atlasvoice.com
canreinvite=no
insecure=very
nat=yes
context=incoming
[6000]
[EMAIL PROTECTED]
[6001]
[6003]
[6004]
[6005]
[6006]
[6007]
[6008]
Thanks,
Pierre
[EMAIL PROTECTED] 2/20/2007 10:47 PM >>>
Hi Pierre,
Just a thought..check your dtmfmode in your SIP client configuration, if
your using inband but your codec is not ulaw or alaw the DTMF tones will be
misrepresented and thus will not be recognised due to the audio compression,
on the other hand if your phones are rfc2833 and asterisk is set to inband
you wont hear anything.
Hope that helps.
Best Regards,
Joanna
On 2/21/07, Pierre Marceau <[EMAIL PROTECTED]> wrote:
Hello,
I can call out to the PSTN and talk to people but when I have to enter a
dtmf tone in an ivr or voicemail system those systems do not recognise that
I have sent a tone. This is the case when I make the call with the Xlite
softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941.
However... a Grandstream GXP2000 works just great ???
All are extensions on my Asterisk 1.4 box. I am using a voip trunk through
Atlasvoice. All extensions are setup identical in sip.conf.
One last thing, if a system wants me to respond 1 for sales 2 for service
I can hit the 1 button quickly 4 or 5 times and the remote system will get
it. That does not work for a three digit extension as you may well imagine.
Any help would be appreciated.
Pierre
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