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Hi all,
Previously I have asked about stopping iLBC in
Asterisk, and I would like to use G.711 u-law only. Actually I have tried
entirely remove anything file related to "ilbc" in /usr/lib/asterisk/modules,
but it still didn't work. The error message about the improper RTP packet length
still there, and I still can't make DTMF detection work.
What's next? Well... thanks to the buggy firmware
and imcompatable standard with Asterisk...
First of all, I can't deny that Planet VIP-450
does a good job in packetizing voice stream, the voice quality is really good
and delay is really small. Also the hardware itself is quite robust, it seldom
halt.. (the machine has been up for a few days). Also it is quite
feature-rich, I can say. BUT I think there is quite a number of BUGS in the
firmware!
In order to see which kind of DTMF Relay it is
using, I have done a packet analysing. When I try to pass SIP INFO type DTMF
band to VIP-450, it replies "501 Unimplemented". Also when I try to pass DTMF
from my POTS phone via the FXO port, only RTP payload can be seen in the packet
captures. I DID suspect that it is RFC2833, because as far as I know RFC2833 did
have the DTMF textx inside the RTP packet somewhere (seems header). But asterisk
just simply did not regconize them (of coz I have set DTMFmode=rfc2833)! It is
pretty strange that the user manual states "VIP handles DTMF Relay per SIP
specification". So VIP-450 actually is using what kind of SIP
specification?
How about using its Inband DTMF relay? This will
certainly generate strange warning just like my case : improper ilbc frame size
and tell me to use u-law to do DTMF even if I AM using G.711 u-law. It is seems
that the DTMF tone generated by VIP-450 generate is kinda strange...
So the final solution is, SIMPLY SWITCH OFF THE DTMF
RELAY IN VIP-450. Please try to type "show coding" in console mode and you will
see a lot of coding (codec) profiles. Most of them are with DTMF relay. Just
switch off them by "set coding <profile id> dtmf_relay off" (please check
with the manual). If you want to stop certain codec, just simply make that
coding profile unusable in voice. For example, "set coding <profile id>
voice off". If you only turn on the profile with u-law, the SIP header
it issues will just consist of 0x4 (ulaw) codec, not 0x105.
In my point of view, Planet
is expecting this device is connected to another VIP-450,
not really for Asterisk or anything else, even not for a soft phone.
Certainly this is not enough for everyone, at least I can't do any IVR and
something what a PBX should have (just like what I can do in Asterisk). I hope
my experience will help anyone who is using VIP-450 with Asterisk, just like me.
I have done Googling for 3 days but I can search for nothing related to this
issue. Sorry for my poor written English.
Cheers,
Jason Chan, Hong Kong
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