[Asterisk-Users] Polycom config and DTMF problems

[Douglas E. Warner]

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    <dwarner at ctinetworks.com>     Network Engineer
CTI Networks, Inc.   http://www.ctinetworks.com    +1 717 975 9000
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