<DTMF tone.dtmf.level="-15" tone.dtmf.onTime="50" tone.dtmf.offTime="50" tone.dtmf.chassis.masking="0" tone.dtmf.stim.pac.offHookOnly="0" tone.dtmf.viaRtp="1" tone.dtmf.rfc2833Control="1" tone.dtmf.rfc2833Payload="101"/>
I'll save you the trouble of checking - it's the same as yours ;-) Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Oct 5, 2005, at 5:02 AM, Douglas E. Warner wrote:
On Tuesday 04 October 2005 18:04, Anthony Rodgers wrote: > I found the best reference to be the SoundPoint IP / SoundStation IP > Admin Guide - SIP 1.5 from the Polycom web site - > http://www.polycom.com/common/pw_item_show_doc/1,1276,4349,00.pdf. >You're right - that admin guide is much more useful that I had initiallythought - thanks! > Not sure about the DTMF issue - I used the config files at > http://www.krisk.org/asterisk/pcom/, if that helpsYeah, I have no idea either. I'm going to try to capture the RTP stream and see if it's being sent inband, but I clearly have my sip.cfg file set torfc2833: <DTMF tone.dtmf.level="-15" tone.dtmf.onTime="50" tone.dtmf.offTime="50" tone.dtmf.chassis.masking="0" tone.dtmf.stim.pac.offHookOnly="0" tone.dtmf.viaRtp="1" tone.dtmf.rfc2833Control="1" tone.dtmf.rfc2833Payload="101" />And I've already tried "dtmfmode=inband" in my asterisk sip.conf, so I'm notsure what's going on. -Doug -- Douglas E. Warner <[EMAIL PROTECTED]> Network Engineer CTI Networks, Inc. http://www.ctinetworks.com +1 717 975 9000 <ATT1964844.dat>_______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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