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. >> >>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 > > > -- The problem with the Future is that it keeps turning into the Present. _______________________________________________ --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
