We have VIP-450FO connectected to Panasonic KX-TEM824's. VIP-450 settings: coding profile for voice = 6 (ulaw), DTMF relay=inband. When I tried to use DISA or IVR on Asterisk I've got a message from Asterisk: bogus frame format "ilbc"..., nothing is works, so, my next step was to hear speaking from analog phone to VoIP phones. Quality of voice was enough, but when I sent DTMF from analog side I'd hear only hrrr-hrrr on the digital side. Problem was that level of DTMF was too small for my gateway FXO ports. When I increase the rxgain parameter for that ports everything is working quite well. Now I can make any call from any side of my VIP-450.

Hi all...
Here we can't do any DTMF relay by VIP-450 too, as stated by Santiago.
It's easier to set it disable any DTMF relay..
however, due to the firmware is a little bit buggy (also problematic in VIP-450 V2)
if you use the coding profile 8, which has no DTMF relay, you will still get errors about DTMF in asterisk.. but this happen only by chance..
If so, just change all profile to disable DTMF relaying.

Also, this devices seems to hard to survive in heavy duty environment..
we have 6 VIP-450 here and if we want to do a IVR using those VIP-450,
it simply bricks after some calls can be made. No calls can be made to/from VIP-450 and Asterisk.
It goes right again after you do a power-cycle..
Here heavy duty means only 5~6 calls per gateway per minutes, and calls are inter-gateway only (did not travel thru Internet)

Just for the record, we had some trouble at first understanding each mode since our hardware (Planet VIP-450) used three possible configurations:

• DTMF Relay disabled = Asterisk inband
• DTMF Relay enabled Inband = Asterisk 2833
• DTMF Relay enabled out of band = Asterisk INFO

I think those names (inband for rfc 2833, out of band for INFO) are normally used so Asterisk names sound a bit confusing at first.

However, we could only get it working by disabling DTMF in VIP and setting Asterisk DTMF mode to inband (luckily we are using ulaw internally). When we tried to use 2833, we got some weird errors like:

• WARNING 31359 : channel.c:2333 set_format: Unable to find a codec translation path from ilbc to ulaw
• WARNING 31359 : chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 1024 (read/write = 4/2)

When ilbc codec module wasn't loaded, or, if it was enabled errors like:

• WARNING 31409 : codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?

When using out of band in hardware and SIP info in Asterisk, tones were sent as Comfort Noise packets from the planet (according to ethereal) and Asterisk obviously generated a warning since it doesn't support comfort noise.

Asterisk version was 'Asterisk 1.2.5 built by bachbuilder @ g77.physik.fu-berlin.de on a i686 running Linux'

This would be a great funciton that we urgently need, but for some reason does not work.

• CLI> sip reload

Reloading SIP
 == Parsing '/etc/asterisk/sip.conf': Found
Oct 21 14:46:19 WARNING19754: chan_sip.c:8367 build_user: Unknown dtmf mode 'auto', using rfc2833
Oct 21 14:46:19 WARNING19754: chan_sip.c:8591 build_peer: Unknown dtmf mode 'auto', using rfc2833
Oct 21 14:46:19 WARNING19754: chan_sip.c:8367 build_user: Unknown dtmf mode 'auto', using rfc2833
Oct 21 14:46:19 WARNING19754: chan_sip.c:8591 build_peer: Unknown dtmf mode 'auto', using rfc2833

• CLI> show version

Asterisk CVS-v1-0-10/21/05-12:31:23 built by root@joro on a i686 running Linux

• CLI>

I compiled this CVS version today, Oct. 21.

We are using only alaw and ulaw for these particular peers.

If anyone knows more about this function and on which version it is supposed to work.

"Also, dtmfmode=info does not work with Asterisks voicemail system."

I'm an * newb so I don't want to edit entries here until I'm more confident but it seems to me that line quoted above is no longer true. I'm running 1.0.3 and using dtmfmode=info for my SIP connections and voicemail works fine.