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 > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
