[Asterisk-Users] Experience sharing on Planet VIP-450 + Asterisk
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 youonly turn on the profile withu-law, the SIP header it issues will just consist of 0x4 (ulaw) codec, not 0x105. In mypoint of view, Planet isexpectingthis deviceisconnected to another VIP-450, not really for Asterisk or anything else, even not fora soft phone. Certainly this is not enough for everyone, at leastI 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 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.1/204 - Release Date: 15/12/2005 ___ --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
RE: [Asterisk-Users] Experience sharing on Planet VIP-450 + Asterisk
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
Re: [Asterisk-Users] Experience sharing on Planet VIP-450 + Asterisk
Jason Chan (jasonOfficial) wrote: 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? Sounds familiar.See below. 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 youonly turn on the profile withu-law, the SIP header it issues will just consist of 0x4 (ulaw) codec, not 0x105. This is what got my attention.Take a look at the commands that I use for the Yoda VG-400. If I'm not mistaken, they're exactly the same for the Planet.Same firmware libraries I presume. http://lists.digium.com/pipermail/asterisk-users/2005-August/120588.html If you notice, I also set dtmf_relay off.Too bad you didn't post any commands earlier, you would have saved a lot of time. Looks like we should compare notes. In mypoint of view, Planet isexpectingthis deviceisconnected to another VIP-450, not really for Asterisk or anything else, even not fora soft phone. Certainly this is not enough for everyone, at leastI 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 This is exactly what Yoda wants as well. I remember planning to buy a Planet unit during my love-hate relationship with Yoda and Asterisk but I'm now glad I didn't. It seems I would have had to tackle the same problems. I'm just happy the units I have work well. Thanks to Yoda's support. They're set on GnuGK rather than Asterisk so it was a first for me as well as them. Xie Xie -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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
