[Asterisk-Users] G.723.1, pass thru and DTMF. Possible?
I am investigating the use of Asterisk for a new project and am confused about all the literature available on G.723.1 in pass thru mode. Specifically, I need to be able to take 2 H.323 channels, each running a G.723.1 codec, and bridge them together. However, before I do that, I need to play a message and then listen to one of the channels to determine how to route the call. For example, it may play a menu asking the user to select one of technical support, sales, or accounting. Or it might ask them to press 1 for Mandarin and route the call to Singapore, 2 for Khmer and and route to Phenom Phen, etc. Since I can program the gateways which will be interfacing to the PSTN to send the DTMF tones via H.245 out of band, there should be no technical reason why this won't work...in other words, I never have to actually decode the G.723.1 stream. The messages can be stored in G.723,1 format already, so I never have to encode either. However, I have not been able to discern whether Asterisk will work in this mode or not. Can someone who has actually implemented an Asterisk system using G.723.1 in pass thru enlighten me? The documentation on this is very confusing. Will it use the out of band H.245 messages to detect a DTMF tone? Since there is no technical reason it should not work, if the answer is it doesn't work can someone give me a hint on what must be changed to make it functional? I would certainly be willing to put in a little effort to make this functional if it isn't already and someone can give me some instructions on where to begin. Thanks in advance for any assistance, Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.723.1 and Asterisk
I have a Cisco ATA 186 working with h323, and G.723.1 codec, but when it makes a connection to a PBX phone, connected to Asterisk by a Digium E100P, don't use G.723.1 codec, the command oh323 show info indicates G.711 for it. Anyone got an idea if Asterisk translates G.723.1 to ISDN channel ? Thanks, Rafael Mayor Rafael Mario Olivieri Comando de Comunicaciones e Informática Dpto Comunicaciones - Jefe Div C4 4346-6137 4346-6100 int 6137 Este mensaje y sus adjuntos son de caracter confidencial para uso de los destinatarios a los que está dirigido. Las opiniones vertidas en este correo son exclusivas de su autor y no representa la opinión del Ejército Argentino. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.723.1 and Asterisk
On Tue, 2004-07-06 at 09:48, [EMAIL PROTECTED] wrote: I have a Cisco ATA 186 working with h323, and G.723.1 codec, but when it makes a connection to a PBX phone, connected to Asterisk by a Digium E100P, don't use G.723.1 codec, the command oh323 show info indicates G.711 for it. Anyone got an idea if Asterisk translates G.723.1 to ISDN channel ? Use google. You will find that the cost of getting G723 implemented in anything is prohibitively expensive due to patents. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.723.1
Hi all, I have a g.723.1 file and my voice devices support this codec, I need to playback this file in asterisk , I stored it in the directory /var/lib/asterisk/sounds/ but when I executte the command in the extension.conf (exten = 100,1,playback(file.g7323) the call hang up, my voice devices are configured with g723 codec, I read that * pass through this codec, so I dont know why this configuration dont work well, if anybody have some idea to respet let me know. I will appreciate you support Best regards Cesar Rico. image001.jpg
Re: [Asterisk-Users] G.723.1
If you don't have the licences for this codec, you can't playback files from *. If I'm not mistaken, * can be used to do codec passthrough between two endpoints, but you can't use any application to interact with *, like voicemail, directory, background or playback. Regards, Gus - Original Message - From: Cesar Rico To: [EMAIL PROTECTED] Sent: Friday, January 23, 2004 7:03 PM Subject: [Asterisk-Users] G.723.1 Hi all, I have a g.723.1 file and my voice devices support this codec, I need to playback this file in asterisk , I stored it in the directory /var/lib/asterisk/sounds/ but when I executte the command in the extension.conf (exten = 100,1,playback(file.g7323) the call hang up, my voice devices are configured with g723 codec, I read that * pass through this codec, so I don't know why this configuration don't work well, if anybody have some idea to respet let me know. I will appreciate you support Best regards Cesar Rico. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.723.1 codec
Had a look at your code and is looking good - need to add it to * Looked for a conversion tool as well for WAV/GSM G.723.1. (could not find lbccodec) Does anybody have a suggestions where to find conversion tools Cheers Dan Andrei Koulik wrote: I solve it for h323 in follow way: 1. Exclude all codecs except g723.1 from h323.conf: disallow=ULAW allow=g723.1 2. Add format_g723 module (http://www.agk.nnov.ru/format_g723.c.gz) into project 3. convert all wav and gsm sound into g723 format (use lbccodec from g723_1 demo package, don't ask me where you can download it) 4. set maxsilence=0 in voicemail.conf to suppress conversion into pcm format for silence detection. And it works fine for me. But where are some bugs in h323 module: * not supported g7231 without sound detection (simple to fix). * sometime data transfer (rtp traffic) begins before negotiation complete and first packet is going in g711 codec and channel going down (not yet reviewed). if will any question regards format_g723 module send mail to: f723 agk.nnov.ru Friday, January 16, 2004, 10:30:41 PM, Dan Tusa wrote: DT Hi, DT Want to do some experiments with the G.723 codecs - where can I download the DT 723 source code for Asterisk? DT I know there are some ongoing discussion regarding patents and license fees DT for the g.723 but I have some hardware on which I only have the 723 and need DT to test it privately. DT Thanks! DT Dan _ Stay in touch with absent friends - get MSN Messenger http://www.msn.co.uk/messenger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.723.1 codec
I solve it for h323 in follow way: 1. Exclude all codecs except g723.1 from h323.conf: disallow=ULAW allow=g723.1 2. Add format_g723 module (http://www.agk.nnov.ru/format_g723.c.gz) into project 3. convert all wav and gsm sound into g723 format (use lbccodec from g723_1 demo package, don't ask me where you can download it) 4. set maxsilence=0 in voicemail.conf to suppress conversion into pcm format for silence detection. And it works fine for me. But where are some bugs in h323 module: * not supported g7231 without sound detection (simple to fix). * sometime data transfer (rtp traffic) begins before negotiation complete and first packet is going in g711 codec and channel going down (not yet reviewed). if will any question regards format_g723 module send mail to: f723 agk.nnov.ru Friday, January 16, 2004, 10:30:41 PM, Dan Tusa wrote: DT Hi, DT Want to do some experiments with the G.723 codecs - where can I download the DT 723 source code for Asterisk? DT I know there are some ongoing discussion regarding patents and license fees DT for the g.723 but I have some hardware on which I only have the 723 and need DT to test it privately. DT Thanks! DT Dan DT _ DT Use MSN Messenger to send music and pics to your friends DT http://www.msn.co.uk/messenger DT ___ DT Asterisk-Users mailing list DT [EMAIL PROTECTED] DT http://lists.digium.com/mailman/listinfo/asterisk-users DT To UNSUBSCRIBE or update options visit: DThttp://lists.digium.com/mailman/listinfo/asterisk-users -- Andrei Koulik. System administrator, Sandy Info Ltd. (ISP), Nizhny Novgorod, Russia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] G.723.1 codec
Saturday, January 17, 2004, 12:49:26 AM, Eric Wieling wrote: EW You can purchase the G.723.1 reference code from the ITU, then you'll EW need to make it work with Asterisk I made codec_g723 with this code, but for compression of PCM file 12 sec long requires 37 sec :) (2x600MHz server) So my opinion is: if server has't any hardware DSP you should not do any codec conversation more complex then(gsm - pcm) EW On Fri, 2004-01-16 at 13:30, Dan Tusa wrote: Hi, Want to do some experiments with the G.723 codecs - where can I download the 723 source code for Asterisk? I know there are some ongoing discussion regarding patents and license fees for the g.723 but I have some hardware on which I only have the 723 and need to test it privately. Thanks! Dan _ Use MSN Messenger to send music and pics to your friends http://www.msn.co.uk/messenger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrei Koulik. System administrator, Sandy Info Ltd. (ISP), Nizhny Novgorod, Russia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.723.1 codec
Hi, Want to do some experiments with the G.723 codecs - where can I download the 723 source code for Asterisk? I know there are some ongoing discussion regarding patents and license fees for the g.723 but I have some hardware on which I only have the 723 and need to test it privately. Thanks! Dan _ Use MSN Messenger to send music and pics to your friends http://www.msn.co.uk/messenger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.723.1 codec
On Friday 16 January 2004 13:30, Dan Tusa wrote: Want to do some experiments with the G.723 codecs - where can I download the 723 source code for Asterisk? I know there are some ongoing discussion regarding patents and license fees for the g.723 but I have some hardware on which I only have the 723 and need to test it privately. Careful. There's a difference between G.723.1 and G.723. The former is patent-encumbered and has no source for Asterisk, while the latter is not patent-encumbered and is known within Asterisk as ADPCM. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.723.1 codec
You can purchase the G.723.1 reference code from the ITU, then you'll need to make it work with Asterisk On Fri, 2004-01-16 at 13:30, Dan Tusa wrote: Hi, Want to do some experiments with the G.723 codecs - where can I download the 723 source code for Asterisk? I know there are some ongoing discussion regarding patents and license fees for the g.723 but I have some hardware on which I only have the 723 and need to test it privately. Thanks! Dan _ Use MSN Messenger to send music and pics to your friends http://www.msn.co.uk/messenger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.723.1
Title: Mensaje Hi, I want to use G.723.1 on *, I read it is supported in Pass Through mode, but I don't understand whats the meaning of that. I have a GW 5300 and an ATA 186 and I want to place calls to PSTN. I setup this config: [general]port = 5060 bindaddr = xx.xx.xx.xx context = sip tos=throughput maxexpirey=360 defaultexpirey=120 [gw5300]type=friendinsecure=yeshost=xx.xx.xx.xxdisallow=allallow=g723allow=ulawcanreinvite=noreinvite=nodtmfmode=rfc2833 [1500]type=friendusername=1500secret=x disallow=allallow=g723allow=ulawhost=dynamiccanreinvite=noqualify=300dtmfmode=rfc2833 and this extension.conf [sip] exten = _0114XXX,1,Dial(SIP/[EMAIL PROTECTED]:5060) where xx.xx.xx.xx is GW ip address but when I place a call from ATA to GW, telephone rings and inmediatly hangs when person answer the phone. When I use only ULAW, all works OK. somebody can tell what I am missing?. someone can help configuring * to use G723 pass through
