Hi Pierre,<br><br>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.
<br><br>Hope that helps.<br><br>Best Regards,<br>Joanna<span style="font-family: monospace;"></span><br><br><div><span class="gmail_quote">On 2/21/07, <b class="gmail_sendername">Pierre Marceau</b> <<a href="mailto:pierre@forestcitynetwerxs.com">
pierre@forestcitynetwerxs.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hello,<br><br>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.
<br><br>However... a Grandstream GXP2000 works just great ???<br><br>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.<br><br>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.
<br><br>Any help would be appreciated.<br><br>Pierre<br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list
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