Hi Pierre, You can also add the following if you think its helpful.
relaxdtmf = yes ;relaxes the DTMF detection parameters qualify = yes ; will send a SIP Optoin command regularly to check if the device is still online, if the device did not answer within 2 seconds it will be considered offline (default time is 2 seconds but can be configured) Best Regards, Joanna On 2/21/07, Benjamin Jacob <[EMAIL PROTECTED]> wrote:
rfc2833 is the prefered way, as inband will work perfectly only with the ulaw codec. 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. >>> >>>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
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