Benjamin Jacob wrote:
rfc2833 is the prefered way, as inband will work perfectly only with the
ulaw codec.
Out of curiosity, there is any 'document' about how RFC2833 would be
better or worse than SIP INFO ?
Pierre Marceau wrote:
Okay, in the SPA-941 admin I changed:
;DTMF Tx Method: Auto
DTMF Tx Method: Inband
and now it works.
Thanks!
Pierre
[EMAIL PROTECTED] 2/21/2007 12:09 AM >>>
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.
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