I have had recent issues with DTMF using Teliax (since their upgrade). Teliax asked me to set my Polycom 501s to SIPINFO for DTMF. They asked that Asterisk and my trunk remain on rfc2833. I can't find any docs on where to set that in the /tftpboot/sip.conf. Does anyone know?
Thanks
-Doug
If I'm understanding correctly you'll want to set tone.dtmf.viaRtp to 0 and tone.dtmf.chassis.masking to 1, this will cause DTMF to be only sent via SIP and not in the audio stream.
-Kale
It appears that makes the phone use inband.
Thanks
-Doug
Let me know what you find out. I was told to do the same thing by Teliax. I'm using IAX and I'm going to give SIP a try before attempting the modify the polycom settings. Let me know how it works out for you and I'll do the same.
Jon
It did not fix the dtmf issues. Although the sound quality improved, so I would recommend it.
-Doug
I have tried the above settings with no luck. Here is some more info from the polycom guide for anyone who is interested.
tone.dtmf.chassis.masking
0, 1
if set to 1, DTMF tones will be substituted with
a non-DTMF pacifier tone when dialing in
hands-free mode to prevent DTMF digits being
broadcast to other surrounding telephony
devices or being inadvertently transmitted in-
band due to local acoustic echo
Note: tone.dtmf.chassis.masking must be
disabled for SIP
Note: chassis.masking should only be enabled
when tone.dtmf.viaRtp is disabled.
tone.dtmf.stim.pac.offHookOnly
0, 1
not currently used
tone.dtmf.viaRtp
0, 1
if set to 1, encode DTMF in the active RTP
stream, otherwise, DTMF may be encoded
within the signaling protocol only (not
supported for SIP)
Note: tone.dtmf.viaRtp must be enabled for
SIP
Note: chassis.masking should be enabled
when tone.dtmf.viaRtp is disable
This did the trick for me, from the Teliax Forums.
"I did quite a bit of looking, and have come to the conclusion that Polycom's do not support SIP-INFO DTMF.
It appears that using inband DTMF and the uLaw CODEC is your only solution. Keep the teliax peer at rfc2833. You could have your local asterisk transcode G729-uLaw provided you have a license and enough horsepower, but I would just use uLaw all around.
A $40 Grandstream BT101 supports SIP-INFO DTMF but Polycoms do not! WOW!
Lonnie"
Try this:
1-Have Teliax set your account to auto for DTMF negotiation.
2-Switch your teliax trunk to SIP if it is not already.
3-Add this line to your teliax sip trunk peer details in FreePBX
dtmfmode=rfc2833
4-Set you extentions in FreePBX to dtmfmode=rfc2833
5-locate the tones section of your /tftpboot/sip.cfg. Change it to match this:
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="0" tone.dtmf.rfc2833Payload="101"
6-Reboot your phones and test.
This should set everything but your phones to rfc2833 and your phones to sipinfo.
Thanks
-Doug
kalleo,
Thanks for the post, this worked for me. Also, the phone number I was calling to test DTMF was Teliax themselves. The rep on the phone stated that it was probably a problem on their end and try another number. That other number worked! After I got it working though I noticed that I no longer could access internal voicemail. Internal DTMF was now broken.
I changed my tones line to: tone.dtmf.rfc2833Control="1" and now internal AND external DTMF works! This may be due to the fact that I am using Polycom 430's and you used 501's so mileage may vary based on the phone model.
Thanks,
Mike
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