<div>Thanks,</div>
<div> </div>
<div>But, I do not have phones connected to Asterisk ...</div>
<div>but only one peer : my softswitch ...</div>
<div>So call flow is Phone -> Softswitch -> Asterisk -> Voicemail </div>
<div> </div>
<div>I can force the link Sofswitch -> Asterisk ( Codec and DMTF Mode )</div>
<div>Codec is PCMx ...</div>
<div>but as i said inband config is not working all the time !</div>
<div> </div>
<div>Let me know if you think something else ...</div>
<div> </div>
<div>JMS<br><br> </div>
<div><span class="gmail_quote">On 2/22/06, <b class="gmail_sendername">Fabian Müller</b> <<a href="mailto:fabian_mueller@open-tk.de">fabian_mueller@open-tk.de</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">"Jean-Marc Salsa" <<a href="mailto:jsalsa@gmail.com">jsalsa@gmail.com</a>> writes:<br><br>
> Which mode should I force into sip.conf ( general, only for peer ? )<br>> so that the Voicemail application is understanding password from users ...<br><br>This depends on what your users are using. If you are using a
<br>Grandstream device you can configure in its administration interface<br>which dtmf mode the telefone should use. If your IP phone is<br>configured to use rfc2833 for example then you would write<br>dtmfmode=rfc2833 in your
sip.conf. If all users use the same<br>dtmfmode it should be ok to write this to the general section.<br><br>Fabian Müller<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">
Easynews.com</a> --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users
</a><br></blockquote></div><br>