> > 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
You should post this stuff and future findings on the wiki. Thanks, Steve _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users