[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
[Asterisk-Users] Anyone with VIP-450
Hi all, While I am making my VoIP gateway to fix G.711, I just want to know who has done Asterisk with Planet VIP-450T DTMF out-of-band relay before... The supporting document says "DTMF relay uses SIP specification", it sounds like it support SIP-INFO, but I'm not sure. Any help will be pleased~ Best Regards, Jason Chan, Hong Kong No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date: 9/12/2005 ___ --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
[Asterisk-Users] I don't want ilbc, i just want G.711
Hi there,I am writing to ask about how to fix the codec to G.711 ONLY.Actually what I am doing is, try to use DTMF when the POTS phone call hasdirected to Asterisk via Planet VIP-450 FXO Port, but this gateway justsimply doesn't support RFC2833 nor SIP-INFO. The only method I can use isInband DTMF. I know it only support G.711, but I DID disallow others andmake it work only with G.711. But the problem is, although I disallow allother codecs, ilbc still itching me...[extensions.conf][852]username=HKGWserect=blahtype=friendhost=dynamicnat =yescanreinvite=nodisallow=alldisallow=ilbcallow=ulawdtmfmode=inband(P.S. I don't use REINVITE simply because I need the asterisk to be amedia gateway cause the gateway is inside NAT behind the Asterisk)Whenever I try to pass DTMF from phone to Asterisk via that gateway, I gotsuch messages:Dec 14 23:35:32 WARNING[10958]: dsp.c:1422 ast_dsp_process: Inband DTMF isnot supported on codec ilbc. Use RFC2833Dec 14 23:35:32 WARNING[10958]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?How come!? I DID DISALLOW them, but it keeps bugging me=192.168.2.3 852 79f9e0-c0a8 00101/1 ulaw No Rx:ACK1 active SIP channel*CLI sip show channel 79 * SIP Call Direction: Incoming Call-ID: [EMAIL PROTECTED] Our Codec Capability: 4 Non-Codec Capability: 0 Their Codec Capability: 261 Joint Codec Capability: 4 Format ulaw Theoretical Address: 192.168.2.3:5060 Received Address: 192.168.2.3:5060 NAT Support: Always Audio IP: 192.168.2.1 (local) Our Tag: as737358ce Their Tag: 3a53f3e1-bbfcafe6d5c SIP User agent: Username: 852 Peername: 852 Original uri: sip:[EMAIL PROTECTED]:5060 Caller-ID: elite Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:[EMAIL PROTECTED]:5060 DTMF Mode: inband SIP Options: (none)==Previously I installed 1.0.3 in same machine, but I overwrite all fileswith 1.2.1.. does it cause a trouble?Can anyone figure out what is the problem? ==Thanks very much for your help!Best regards,Jason Chan, Hong Kong No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date: 9/12/2005 ___ --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
