Thanks for the answer ;) I'm using H323 (the one that comes under /channels/h323 with asterisk source). Before upgrading asterisk I prefer to try what you say (using OH323).
Do you know which one is better? OH323 or ooH323c? (the second comes with asterisk-addons) Thanks again. Alejandro -----Mensaje original----- De: Oliver Vermeulen [mailto:[EMAIL PROTECTED] Enviado el: Wednesday, April 19, 2006 4:09 PM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' CC: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Codec problem from SIP to H323 Try to upgrade asterisk to version 1.2.4 Are you using OH323 or H323 ? I had same problem with 1.2.1 using H323(addon) , Installed 1.2.4 and OH323 and everything worked fine. Cheers, Oliver -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Mejía Evertsz Sent: Thursday, April 20, 2006 12:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Codec problem from SIP to H323 Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf has "disallow=all & allow=g729" - h323.conf has "disallow=all & allow=g729" The problem: [SIPphone] [sip.conf] [h323.conf] [H323gw] g729 ---> allow=g729 ---> allow=g729 ---> g729 When I dial to the gateway from the SIPphone using g729 as my sip phone's default codec I get: -- Executing Dial("SIP/amejia-8be1", "H323/[EMAIL PROTECTED]") in new stack Apr 19 15:02:14 WARNING[68595]: channel.c:2504 ast_request: No translator path exists for channel type H323 (native 4) to 256 Apr 19 15:02:14 NOTICE[68595]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'H323' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) I don't get it why is it trying to "translate" anything. There's nothing to translate, cause I'm using g729 in both ends. Well, to make it more interesting, I tried this way: [SIPphone] [sip.conf] [h323.conf] [H323gw] g711 ---> allow=all ---> allow=all ---> g729 This way, it passes the call to the gateway just giving a waring that it can't find a codec to translate. But at least it passes the call. It rings on the other side, and of course as I don't have any g729 licenses installed it drops the call when answered. -- Executing Dial("SIP/amejia-1fc8", "H323/[EMAIL PROTECTED]") in new stack -- Called [EMAIL PROTECTED] Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw -- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8 -- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8 -- H323/H323gw-2 is ringing -- H323/H323gw-2 answered SIP/amejia-1fc8 Apr 19 15:23:45 WARNING[75484]: channel.c:2685 ast_channel_make_compatible: No path to translate from SIP/amejia-1fc8(4) to H323/H323gw-2(256) Apr 19 15:23:45 WARNING[75484]: app_dial.c:1553 dial_exec_full: Had to drop call because I couldn't make SIP/amejia-1fc8 compatible with H323/H323gw-2 == Spawn extension (test, 444, 1) exited non-zero on 'SIP/amejia-1fc8' Does anybody know how can I get rid of the problem I get on the first scenario? Why does it try to use codec 4 (g711u) if both ends are configured with g729? Please give me some light. I don't know what else to try. Thank you all. Alejandro Mejia _______________________________________________ --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 _______________________________________________ --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
