hello i was searching for solution to problem (sip->h.323). any one from this list asterisk mailing have any idea how to fix it.
i am getting error when i try to call from sip to h.323 user i am successfully registering my asterisk box with gnugk. but when i try to call to h.323 openphone on working on GnuGatekeeper, asterisk is not routing it to GnuGk. i am getting the following error. do you have any idea. please help i am stuck here for a week. i am unable to find anything on google on this topic. -- Executing Dial("SIP/2000-ae3f", "OH323/[EMAIL PROTECTED]:1720") in new stack Mar 16 16:14:46 ERROR[16176]: chan_oh323.c:2501 ast_oh323_new: Internal channel initialization failed. Bad binary? Mar 16 16:14:46 WARNING[16176]: chan_oh323.c:2727 oh323_request: Failed to create new H.323 private structure 1. Mar 16 16:14:46 NOTICE[16176]: app_dial.c:749 dial_exec: Unable to create channel of type 'OH323' == Everyone is busy/congested at this time Mar 16 16:20:55 WARNING[16176]: res_musiconhold.c:205 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Mar 16 16:20:55 WARNING[16176]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player thanks kamran __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ _______________________________________________ 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