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 initially thought - thanks! > Not sure about the DTMF issue - I used the config files at > http://www.krisk.org/asterisk/pcom/, if that helps Yeah, 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 to rfc2833: <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 not sure what's going on. -Doug -- Douglas E. Warner <[EMAIL PROTECTED]> Network Engineer CTI Networks, Inc. http://www.ctinetworks.com +1 717 975 9000
pgpSma0yCYIDV.pgp
Description: PGP signature
_______________________________________________ --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
